From miconda at gmail.com  Wed Aug  1 09:57:36 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Aug 2012 09:57:36 +0200
Subject: [SR-Users] Core fm_realloc function
In-Reply-To: <CAKv6MrZxGYHCTUr3ogtHJJqVYMig1DN7yokMmFHpUkjBPt7cpw@mail.gmail.com>
References: <CAKv6MrYkEVCs8taSaEXNcV2ohi6yQSLHMAWpAqEc_FK4HAhXFw@mail.gmail.com>
	<50059EC2.6060409@gmail.com>
	<CAKv6MrY5_+mJ04pvXFfeYMa8HPSSjujEihjRYzu1azzHC+DR6g@mail.gmail.com>
	<5007089F.708@digitro.com.br>
	<CAKv6Mrauh-2xA4f4H7JMKpUXUFtw+HUJsurwJ8DV+P8v5whmhA@mail.gmail.com>
	<CAKv6MrbFb6krEmLiyHWB0Zho2bPOJ3_fSno945An=55WDm42Gw@mail.gmail.com>
	<50163CE8.2050903@gmail.com>
	<CAKv6MrYOx1iGtUjsiNQwnUnmyBm3m=97BMnGhCVhhgB2gB+mnQ@mail.gmail.com>
	<5016AA13.9030806@gmail.com>
	<CAKv6Mrb_rO-Lr9t9nvb+RvZfzOHQuNpb5uvnNK1ho_jdOLw24Q@mail.gmail.com>
	<5016C320.5040402@gmail.com>
	<CAKv6MrYB1kYTuiAKuJNRiEh4rw_AFU2POP4GD+BHD5h9_ru81A@mail.gmail.com>
	<50177A05.5020104@gmail.com>
	<CAKv6MrZxGYHCTUr3ogtHJJqVYMig1DN7yokMmFHpUkjBPt7cpw@mail.gmail.com>
Message-ID: <5018E170.9040405@gmail.com>

Hello,

On 7/31/12 3:39 PM, Bruno Bresciani wrote:
> Hello,
>
> I didn't know that log messages related to memory operations can be 
> controlled by global parameter, but I like to know if is recommendable 
> I recompile kamailio using q_malloc (default) and not f_malloc... 
> memory operations using q_malloc is more reliable and avoid problems 
> or crashes or it is only more suitable for debugging? What do you 
> suggest Daniel?

q_malloc is more suitable for debugging.

>
> The 'get_statistics all' command is avaliable by a specific module? I 
> run 'kamctl fifo get_statistics all' and return '500 command 
> 'get_statistics' not available'

Do you have kex module loaded? What is the output of 'kamctl fifo which'?

>
> For while is impossible to start a new installation, first because I 
> don't know how much time I will spend to port and second because I am 
> involved with other developments and I have no time  to make this. I 
> know that 3.1 is no longer a official branch but now start a new 
> installation it's very very difficult, my in intention is discover 
> what caused the crash and if exists a way to fix or prevent it.
Discovering may require additional patches, like more debug messages in 
the C code, that's why is better to start with the latest stable.

Cheers,
Daniel

>
> Best Regards
>
> 2012/7/31 Daniel-Constantin Mierla <miconda at gmail.com 
> <mailto:miconda at gmail.com>>
>
>     Hello,
>
>
>     On 7/30/12 9:01 PM, Bruno Bresciani wrote:
>>     Hi,
>>
>>     I compiled kamailio with MEMDBG = 0 because I didn't want the
>>     memory debug  in kamailio log, but I didn't know these turns on
>>     f_malloc and disabling q_malloc...
>     the log messages related to memory operations can be controlled by
>     global parameters memdbg and memlog.
>
>
>>
>>     Probably the size of shared memory that I start kamailio is 32MB
>>     because I didn't gave a different -m parameter value, exist a
>>     command to verify this information?
>
>     kamctl fifo get_statistics all
>
>     and see the shared memory total value. It will be interesting to
>     see available shared memory as well.
>
>
>>
>>     At moment that crash happened, there were few registered users
>>     agents and were being made tests with register and calls with TLS
>>     protocol. I got only the two situations that I showed at first email.
>
>     If you plan to start a new installation, I strongly recommend 3.3
>     branch, the code is more actual and easier to debug. 3.1 is no
>     longer an official maintained branch, those being now 3.3 and 3.2.
>     I'm looking at this issue to be sure it is no longer in latest stable.
>
>     Cheers,
>     Daniel
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From miconda at gmail.com  Wed Aug  1 09:59:21 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Aug 2012 09:59:21 +0200
Subject: [SR-Users] cannot allocate memory & increase shared memory
In-Reply-To: <COL104-W339B4C7B7571AA6D51FA8B84C50@phx.gbl>
References: <COL104-W339B4C7B7571AA6D51FA8B84C50@phx.gbl>
Message-ID: <5018E1D9.2000509@gmail.com>

Hello,

what version are you running?

On 7/31/12 3:06 PM, Alex Solt wrote:
> Hi,
>
> We are getting "cannot allocate memory" error in the log file. I was 
> wondering which one is the right way to increase the shared memory:
>
> 1) adding the following to /etc/openser/openserctl
>
> STARTOPTIONS="-m 128"
this is used when you start via ctl tool.

>
>
> 2) modify the following in /etc/default/openser
>
> MEMORY=1024
this is used when you start via init.d script.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From danb.lists at gmail.com  Wed Aug  1 12:05:32 2012
From: danb.lists at gmail.com (DanB)
Date: Wed, 01 Aug 2012 12:05:32 +0200
Subject: [SR-Users] Segfault running 3.3.0 with registrar/p_usrloc/path
In-Reply-To: <mailman.3.1343815201.12643.sr-users@lists.sip-router.org>
References: <mailman.3.1343815201.12643.sr-users@lists.sip-router.org>
Message-ID: <5018FF6C.9060901@gmail.com>

Hey Guys,

Just a quick follow up to see if anything happened with p_usrloc fixup. 
Any chance to expect it running on new 3.3.1 release?

Ta,
DanB


From marius.zbihlei at 1and1.ro  Wed Aug  1 12:21:58 2012
From: marius.zbihlei at 1and1.ro (Marius Zbihlei)
Date: Wed, 1 Aug 2012 13:21:58 +0300
Subject: [SR-Users] Segfault running 3.3.0 with registrar/p_usrloc/path
In-Reply-To: <5018FF6C.9060901@gmail.com>
References: <mailman.3.1343815201.12643.sr-users@lists.sip-router.org>
	<5018FF6C.9060901@gmail.com>
Message-ID: <50190346.9060604@1and1.ro>

On 08/01/2012 01:05 PM, DanB wrote:
> Hey Guys,
>
> Just a quick follow up to see if anything happened with p_usrloc fixup.
> Any chance to expect it running on new 3.3.1 release?
Hello Dan,

At the moment the GRUU/outbound patch is integrated into the master 
branch([1]). I don't know exactly if this is a bug (so it will be 
cherry-picked to 3.3.x) or a feature, that will be present in upcoming 
3.4. IMHO, I will treat this as a bug, so I will cherry-pick the patch 
and apply it to 3.3.x as well. If somebody objects please let's comment 
on the reasons.

Cheers,
Marius

[1] 
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1f6bfa0b3ba15201c2ca3e2387a9f9e81e989643 

> Ta,
> DanB
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Zbihlei Marius

Head of
Linux Development Services Romania

1&1 Internet Development srl    Tel KA: 754-9152
Str Mircea Eliade 18            Tel RO: +40-31-223-9152
Sect 1, Bucuresti               mailto: marius.zbihlei at 1and1.ro
71295, Romania



From miconda at gmail.com  Wed Aug  1 13:43:49 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Aug 2012 13:43:49 +0200
Subject: [SR-Users] Segfault running 3.3.0 with registrar/p_usrloc/path
In-Reply-To: <50190346.9060604@1and1.ro>
References: <mailman.3.1343815201.12643.sr-users@lists.sip-router.org>
	<5018FF6C.9060901@gmail.com> <50190346.9060604@1and1.ro>
Message-ID: <50191675.1070905@gmail.com>

Hello,

it's a bug, because it does not work otherwise. Go ahead and backport.

Cheers,
Daniel

On 8/1/12 12:21 PM, Marius Zbihlei wrote:
> On 08/01/2012 01:05 PM, DanB wrote:
>> Hey Guys,
>>
>> Just a quick follow up to see if anything happened with p_usrloc fixup.
>> Any chance to expect it running on new 3.3.1 release?
> Hello Dan,
>
> At the moment the GRUU/outbound patch is integrated into the master 
> branch([1]). I don't know exactly if this is a bug (so it will be 
> cherry-picked to 3.3.x) or a feature, that will be present in upcoming 
> 3.4. IMHO, I will treat this as a bug, so I will cherry-pick the patch 
> and apply it to 3.3.x as well. If somebody objects please let's 
> comment on the reasons.
>
> Cheers,
> Marius
>
> [1] 
> http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1f6bfa0b3ba15201c2ca3e2387a9f9e81e989643 
>
>> Ta,
>> DanB
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From gnugk at telegroup.com.ua  Wed Aug  1 17:33:27 2012
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Wed, 01 Aug 2012 18:33:27 +0300
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
Message-ID: <50194C47.1030602@telegroup.com.ua>

The same trouble with:
completely updated Centos 5
the last Kamailio RPM from
http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/
and
postgresql-libs.x86_64.8.1.23-5.el5_8

several month ago I try
postgresql84-libs.x86_64
with the same result.

It's not possible to use Kamailio RPM with postgres backend. Need to 
assemble it manually.


Bruno Bresciani wrote:
 > Hi,
 >
 > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the
 > log file is generated the following error:
 >
 > //
 >
 > ERROR: load_module: could not open module
 > </home2/local/kamailio/lib/kamailio/modules/db_
 > postgres.so>: /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
 > undefined symbol: PQdescribePrepared

Have you built Kamailio yourself? Looks like db_postgres.so can not find
the postgresql libraries, or it was built with a different library version.

Maybe ldd can give you some details:

    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so

regards
Klaus



From miconda at gmail.com  Wed Aug  1 22:00:14 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Aug 2012 22:00:14 +0200
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <50194C47.1030602@telegroup.com.ua>
References: <50194C47.1030602@telegroup.com.ua>
Message-ID: <50198ACE.7030909@gmail.com>

Hello,

can you paste here the output of following command executed in the 
db_postgres module directory:

make Q=0

First run 'make proper' in the same directory. I want to see the compile 
flags and linked libs used for your system.

Cheers,
Daniel

On 8/1/12 5:33 PM, Andrew O. Zhukov wrote:
> The same trouble with:
> completely updated Centos 5
> the last Kamailio RPM from
> http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/ 
>
> and
> postgresql-libs.x86_64.8.1.23-5.el5_8
>
> several month ago I try
> postgresql84-libs.x86_64
> with the same result.
>
> It's not possible to use Kamailio RPM with postgres backend. Need to 
> assemble it manually.
>
>
> Bruno Bresciani wrote:
> > Hi,
> >
> > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the
> > log file is generated the following error:
> >
> > //
> >
> > ERROR: load_module: could not open module
> > </home2/local/kamailio/lib/kamailio/modules/db_
> > postgres.so>: 
> /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
> > undefined symbol: PQdescribePrepared
>
> Have you built Kamailio yourself? Looks like db_postgres.so can not find
> the postgresql libraries, or it was built with a different library 
> version.
>
> Maybe ldd can give you some details:
>
>    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so
>
> regards
> Klaus
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From infoananthk at gmail.com  Thu Aug  2 01:54:03 2012
From: infoananthk at gmail.com (Ananth Kollipara)
Date: Thu, 2 Aug 2012 05:24:03 +0530
Subject: [SR-Users] Regarding - Add your own module
Message-ID: <CACJ9FaubGDLbTRgDHF-nCKL_jjHKDe1Ud1ddrF7GpcqnqaiX9Q@mail.gmail.com>

Hi Alex,

Thanks for the response. Yes, i meant developing of a custom module and
integrating with ser/kamailio. I am looking for the following.

1) On receiving 302 response, proxy shall use the contact header and
re-send the msg to the correct destination.
2) Integration of kamailio with Times Ten database (query Times ten DB)

Please advice.

Regards,
Ananth


On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request at lists.sip-router.org>wrote:

> Send sr-users mailing list submissions to
>         sr-users at lists.sip-router.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> or, via email, send a message with subject or body 'help' to
>         sr-users-request at lists.sip-router.org
>
> You can reach the person managing the list at
>         sr-users-owner at lists.sip-router.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of sr-users digest..."
>
>
> Today's Topics:
>
>    1. Add Own Plug-in (Ananth Kollipara)
>    2. Re: Add Own Plug-in (Alex Balashov)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 31 Jul 2012 14:33:52 +0530
> From: Ananth Kollipara <infoananthk at gmail.com>
> Subject: [SR-Users] Add Own Plug-in
> To: sr-users at lists.sip-router.org
> Message-ID:
>         <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS=
> ZQ at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> Can you please provide the steps to add a plug-in?
>
> Regards,
> Ananth
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>
> ------------------------------
>
> Message: 2
> Date: Tue, 31 Jul 2012 05:10:36 -0400
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [SR-Users] Add Own Plug-in
> To: sr-users at lists.sip-router.org
> Message-ID: <5017A10C.2020107 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 07/31/2012 05:03 AM, Ananth Kollipara wrote:
>
> > Can you please provide the steps to add a plug-in?
>
> You mean develop a custom module for Kamailio?
>
> You'd want to start here:
>
> http://www.asipto.com/pub/kamailio-devel-guide/
>
> However, perhaps a custom module is not the fastest or most optimal way
> to achieve what you want.  There may already be facilities to
> accommodate your requirements.  So, a useful start to the conversation
> might be: what do you want to accomplish?
>
> -- Alex
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>
>
> ------------------------------
>
> _______________________________________________
> sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> End of sr-users Digest, Vol 86, Issue 85
> ****************************************
>
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From abalashov at evaristesys.com  Thu Aug  2 01:59:42 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 01 Aug 2012 19:59:42 -0400
Subject: [SR-Users] Regarding - Add your own module
Message-ID: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>


Does TimesTen have a UnixODBC connector? If so, you can use existing Kamailio modules.?



-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Ananth Kollipara <infoananthk at gmail.com> wrote:Hi Alex,

Thanks for the response. Yes, i meant developing of a custom module and integrating with ser/kamailio. I am looking for the following.

1) On receiving 302 response, proxy shall use the contact header and re-send the msg to the correct destination.
2) Integration of kamailio with Times Ten database (query Times ten DB)

Please advice.

Regards,
Ananth


On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request at lists.sip-router.org> wrote:
Send sr-users mailing list submissions to
? ? ? ? sr-users at lists.sip-router.org

To subscribe or unsubscribe via the World Wide Web, visit
? ? ? ? http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
or, via email, send a message with subject or body 'help' to
? ? ? ? sr-users-request at lists.sip-router.org

You can reach the person managing the list at
? ? ? ? sr-users-owner at lists.sip-router.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of sr-users digest..."


Today's Topics:

? ?1. Add Own Plug-in (Ananth Kollipara)
? ?2. Re: Add Own Plug-in (Alex Balashov)


----------------------------------------------------------------------

Message: 1
Date: Tue, 31 Jul 2012 14:33:52 +0530
From: Ananth Kollipara <infoananthk at gmail.com>
Subject: [SR-Users] Add Own Plug-in
To: sr-users at lists.sip-router.org
Message-ID:
? ? ? ? <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS=ZQ at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

Can you please provide the steps to add a plug-in?

Regards,
Ananth
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------------------------------

Message: 2
Date: Tue, 31 Jul 2012 05:10:36 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [SR-Users] Add Own Plug-in
To: sr-users at lists.sip-router.org
Message-ID: <5017A10C.2020107 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 07/31/2012 05:03 AM, Ananth Kollipara wrote:

> Can you please provide the steps to add a plug-in?

You mean develop a custom module for Kamailio?

You'd want to start here:

http://www.asipto.com/pub/kamailio-devel-guide/

However, perhaps a custom module is not the fastest or most optimal way
to achieve what you want. ?There may already be facilities to
accommodate your requirements. ?So, a useful start to the conversation
might be: what do you want to accomplish?

-- Alex

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/



------------------------------

_______________________________________________
sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


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****************************************

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From infoananthk at gmail.com  Thu Aug  2 02:09:17 2012
From: infoananthk at gmail.com (Ananth Kollipara)
Date: Thu, 2 Aug 2012 05:39:17 +0530
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
Message-ID: <CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>

Yes, it does. What about dynamic routing?

Regards,
Ananth

On Thu, Aug 2, 2012 at 5:29 AM, Alex Balashov <abalashov at evaristesys.com>wrote:

>
> Does TimesTen have a UnixODBC connector? If so, you can use existing
> Kamailio modules.
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> Ananth Kollipara <infoananthk at gmail.com> wrote:
> Hi Alex,
>
> Thanks for the response. Yes, i meant developing of a custom module and
> integrating with ser/kamailio. I am looking for the following.
>
> 1) On receiving 302 response, proxy shall use the contact header and
> re-send the msg to the correct destination.
> 2) Integration of kamailio with Times Ten database (query Times ten DB)
>
> Please advice.
>
> Regards,
> Ananth
>
>
> On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request at lists.sip-router.org>wrote:
>
>> Send sr-users mailing list submissions to
>>         sr-users at lists.sip-router.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> or, via email, send a message with subject or body 'help' to
>>         sr-users-request at lists.sip-router.org
>>
>> You can reach the person managing the list at
>>         sr-users-owner at lists.sip-router.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of sr-users digest..."
>>
>>
>> Today's Topics:
>>
>>    1. Add Own Plug-in (Ananth Kollipara)
>>    2. Re: Add Own Plug-in (Alex Balashov)
>>
>>
>> ----------------------------------------------------------------------
>>
>> Message: 1
>> Date: Tue, 31 Jul 2012 14:33:52 +0530
>> From: Ananth Kollipara <infoananthk at gmail.com>
>> Subject: [SR-Users] Add Own Plug-in
>> To: sr-users at lists.sip-router.org
>> Message-ID:
>>         <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS=
>> ZQ at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi,
>>
>> Can you please provide the steps to add a plug-in?
>>
>> Regards,
>> Ananth
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: <
>> http://lists.sip-router.org/pipermail/sr-users/attachments/20120731/7311abed/attachment.html
>> >
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Tue, 31 Jul 2012 05:10:36 -0400
>> From: Alex Balashov <abalashov at evaristesys.com>
>> Subject: Re: [SR-Users] Add Own Plug-in
>> To: sr-users at lists.sip-router.org
>> Message-ID: <5017A10C.2020107 at evaristesys.com>
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>> On 07/31/2012 05:03 AM, Ananth Kollipara wrote:
>>
>> > Can you please provide the steps to add a plug-in?
>>
>> You mean develop a custom module for Kamailio?
>>
>> You'd want to start here:
>>
>> http://www.asipto.com/pub/kamailio-devel-guide/
>>
>> However, perhaps a custom module is not the fastest or most optimal way
>> to achieve what you want.  There may already be facilities to
>> accommodate your requirements.  So, a useful start to the conversation
>> might be: what do you want to accomplish?
>>
>> -- Alex
>>
>> --
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 235 E Ponce de Leon Ave
>> Suite 106
>> Decatur, GA 30030
>> Tel: +1-678-954-0670
>> Fax: +1-404-961-1892
>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> End of sr-users Digest, Vol 86, Issue 85
>> ****************************************
>>
>
>
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From abalashov at evaristesys.com  Thu Aug  2 02:11:06 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 01 Aug 2012 20:11:06 -0400
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
Message-ID: <5019C59A.7000301@evaristesys.com>


What about it?

On 08/01/2012 08:09 PM, Ananth Kollipara wrote:

> Yes, it does. What about dynamic routing?
>
> Regards,
> Ananth
>
> On Thu, Aug 2, 2012 at 5:29 AM, Alex Balashov <abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>> wrote:
>
>
>     Does TimesTen have a UnixODBC connector? If so, you can use existing
>     Kamailio modules.
>
>
>
>     -- Alex
>
>     --
>     Sent from my Samsung mobile, and thus lacking in the refinement one
>     might expect from a proper keyboard.
>
>     Alex Balashov - Principal
>     Evariste Systems LLC
>     235 E Ponce de Leon Ave
>     Suite 106
>     Decatur, GA 30030
>     Tel: +1-678-954-0670
>     Web: http://www.evaristesys.com/
>
>     Ananth Kollipara <infoananthk at gmail.com
>     <mailto:infoananthk at gmail.com>> wrote:
>     Hi Alex,
>
>     Thanks for the response. Yes, i meant developing of a custom module
>     and integrating with ser/kamailio. I am looking for the following.
>
>     1) On receiving 302 response, proxy shall use the contact header and
>     re-send the msg to the correct destination.
>     2) Integration of kamailio with Times Ten database (query Times ten DB)
>
>     Please advice.
>
>     Regards,
>     Ananth
>
>
>     On Tue, Jul 31, 2012 at 3:30 PM,
>     <sr-users-request at lists.sip-router.org
>     <mailto:sr-users-request at lists.sip-router.org>> wrote:
>
>         Send sr-users mailing list submissions to
>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>
>         To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         or, via email, send a message with subject or body 'help' to
>         sr-users-request at lists.sip-router.org
>         <mailto:sr-users-request at lists.sip-router.org>
>
>         You can reach the person managing the list at
>         sr-users-owner at lists.sip-router.org
>         <mailto:sr-users-owner at lists.sip-router.org>
>
>         When replying, please edit your Subject line so it is more specific
>         than "Re: Contents of sr-users digest..."
>
>
>         Today's Topics:
>
>             1. Add Own Plug-in (Ananth Kollipara)
>             2. Re: Add Own Plug-in (Alex Balashov)
>
>
>         ----------------------------------------------------------------------
>
>         Message: 1
>         Date: Tue, 31 Jul 2012 14:33:52 +0530
>         From: Ananth Kollipara <infoananthk at gmail.com
>         <mailto:infoananthk at gmail.com>>
>         Subject: [SR-Users] Add Own Plug-in
>         To: sr-users at lists.sip-router.org
>         <mailto:sr-users at lists.sip-router.org>
>         Message-ID:
>
>         <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS=ZQ at mail.gmail.com
>         <mailto:ZQ at mail.gmail.com>>
>         Content-Type: text/plain; charset="iso-8859-1"
>
>         Hi,
>
>         Can you please provide the steps to add a plug-in?
>
>         Regards,
>         Ananth
>         -------------- next part --------------
>         An HTML attachment was scrubbed...
>         URL:
>         <http://lists.sip-router.org/pipermail/sr-users/attachments/20120731/7311abed/attachment.html>
>
>         ------------------------------
>
>         Message: 2
>         Date: Tue, 31 Jul 2012 05:10:36 -0400
>         From: Alex Balashov <abalashov at evaristesys.com
>         <mailto:abalashov at evaristesys.com>>
>         Subject: Re: [SR-Users] Add Own Plug-in
>         To: sr-users at lists.sip-router.org
>         <mailto:sr-users at lists.sip-router.org>
>         Message-ID: <5017A10C.2020107 at evaristesys.com
>         <mailto:5017A10C.2020107 at evaristesys.com>>
>         Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>         On 07/31/2012 05:03 AM, Ananth Kollipara wrote:
>
>          > Can you please provide the steps to add a plug-in?
>
>         You mean develop a custom module for Kamailio?
>
>         You'd want to start here:
>
>         http://www.asipto.com/pub/kamailio-devel-guide/
>
>         However, perhaps a custom module is not the fastest or most
>         optimal way
>         to achieve what you want.  There may already be facilities to
>         accommodate your requirements.  So, a useful start to the
>         conversation
>         might be: what do you want to accomplish?
>
>         -- Alex
>
>         --
>         Alex Balashov - Principal
>         Evariste Systems LLC
>         235 E Ponce de Leon Ave
>         Suite 106
>         Decatur, GA 30030
>         Tel: +1-678-954-0670
>         Fax: +1-404-961-1892
>         Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>
>
>         ------------------------------
>
>         _______________________________________________
>         sr-users mailing list
>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>         End of sr-users Digest, Vol 86, Issue 85
>         ****************************************
>
>
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From infoananthk at gmail.com  Thu Aug  2 02:12:32 2012
From: infoananthk at gmail.com (Ananth Kollipara)
Date: Thu, 2 Aug 2012 05:42:32 +0530
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <5019C59A.7000301@evaristesys.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
Message-ID: <CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>

Which module does dynamic routing?

On Thu, Aug 2, 2012 at 5:41 AM, Alex Balashov <abalashov at evaristesys.com>wrote:

>
> What about it?
>
>
> On 08/01/2012 08:09 PM, Ananth Kollipara wrote:
>
>  Yes, it does. What about dynamic routing?
>>
>> Regards,
>> Ananth
>>
>> On Thu, Aug 2, 2012 at 5:29 AM, Alex Balashov <abalashov at evaristesys.com
>> <mailto:abalashov at evaristesys.**com <abalashov at evaristesys.com>>> wrote:
>>
>>
>>     Does TimesTen have a UnixODBC connector? If so, you can use existing
>>     Kamailio modules.
>>
>>
>>
>>     -- Alex
>>
>>     --
>>     Sent from my Samsung mobile, and thus lacking in the refinement one
>>     might expect from a proper keyboard.
>>
>>     Alex Balashov - Principal
>>     Evariste Systems LLC
>>     235 E Ponce de Leon Ave
>>     Suite 106
>>     Decatur, GA 30030
>>     Tel: +1-678-954-0670
>>     Web: http://www.evaristesys.com/
>>
>>     Ananth Kollipara <infoananthk at gmail.com
>>     <mailto:infoananthk at gmail.com>**> wrote:
>>     Hi Alex,
>>
>>     Thanks for the response. Yes, i meant developing of a custom module
>>     and integrating with ser/kamailio. I am looking for the following.
>>
>>     1) On receiving 302 response, proxy shall use the contact header and
>>     re-send the msg to the correct destination.
>>     2) Integration of kamailio with Times Ten database (query Times ten
>> DB)
>>
>>     Please advice.
>>
>>     Regards,
>>     Ananth
>>
>>
>>     On Tue, Jul 31, 2012 at 3:30 PM,
>>     <sr-users-request at lists.sip-**router.org<sr-users-request at lists.sip-router.org>
>>     <mailto:sr-users-request@**lists.sip-router.org<sr-users-request at lists.sip-router.org>>>
>> wrote:
>>
>>         Send sr-users mailing list submissions to
>>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-**
>> router.org <sr-users at lists.sip-router.org>>
>>
>>
>>         To subscribe or unsubscribe via the World Wide Web, visit
>>         http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>         or, via email, send a message with subject or body 'help' to
>>         sr-users-request at lists.sip-**router.org<sr-users-request at lists.sip-router.org>
>>         <mailto:sr-users-request@**lists.sip-router.org<sr-users-request at lists.sip-router.org>
>> >
>>
>>
>>         You can reach the person managing the list at
>>         sr-users-owner at lists.sip-**router.org<sr-users-owner at lists.sip-router.org>
>>         <mailto:sr-users-owner at lists.**sip-router.org<sr-users-owner at lists.sip-router.org>
>> >
>>
>>
>>         When replying, please edit your Subject line so it is more
>> specific
>>         than "Re: Contents of sr-users digest..."
>>
>>
>>         Today's Topics:
>>
>>             1. Add Own Plug-in (Ananth Kollipara)
>>             2. Re: Add Own Plug-in (Alex Balashov)
>>
>>
>>         ------------------------------**------------------------------**
>> ----------
>>
>>         Message: 1
>>         Date: Tue, 31 Jul 2012 14:33:52 +0530
>>         From: Ananth Kollipara <infoananthk at gmail.com
>>         <mailto:infoananthk at gmail.com>**>
>>
>>         Subject: [SR-Users] Add Own Plug-in
>>         To: sr-users at lists.sip-router.org
>>         <mailto:sr-users at lists.sip-**router.org<sr-users at lists.sip-router.org>
>> >
>>         Message-ID:
>>
>>         <CACJ9FatODfYWEG+y6_U_=-**xxsxt65Ca=5FV2NFBQrYw+grS=ZQ@**
>> mail.gmail.com <ZQ at mail.gmail.com>
>>         <mailto:ZQ at mail.gmail.com>>
>>
>>         Content-Type: text/plain; charset="iso-8859-1"
>>
>>         Hi,
>>
>>         Can you please provide the steps to add a plug-in?
>>
>>         Regards,
>>         Ananth
>>         -------------- next part --------------
>>         An HTML attachment was scrubbed...
>>         URL:
>>         <http://lists.sip-router.org/**pipermail/sr-users/**
>> attachments/20120731/7311abed/**attachment.html<http://lists.sip-router.org/pipermail/sr-users/attachments/20120731/7311abed/attachment.html>
>> >
>>
>>         ------------------------------
>>
>>         Message: 2
>>         Date: Tue, 31 Jul 2012 05:10:36 -0400
>>         From: Alex Balashov <abalashov at evaristesys.com
>>         <mailto:abalashov at evaristesys.**com <abalashov at evaristesys.com>>>
>>
>>         Subject: Re: [SR-Users] Add Own Plug-in
>>         To: sr-users at lists.sip-router.org
>>         <mailto:sr-users at lists.sip-**router.org<sr-users at lists.sip-router.org>
>> >
>>         Message-ID: <5017A10C.2020107 at evaristesys.**com<5017A10C.2020107 at evaristesys.com>
>>         <mailto:5017A10C.2020107@**evaristesys.com<5017A10C.2020107 at evaristesys.com>
>> >>
>>
>>         Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>>         On 07/31/2012 05:03 AM, Ananth Kollipara wrote:
>>
>>          > Can you please provide the steps to add a plug-in?
>>
>>         You mean develop a custom module for Kamailio?
>>
>>         You'd want to start here:
>>
>>         http://www.asipto.com/pub/**kamailio-devel-guide/<http://www.asipto.com/pub/kamailio-devel-guide/>
>>
>>         However, perhaps a custom module is not the fastest or most
>>         optimal way
>>         to achieve what you want.  There may already be facilities to
>>         accommodate your requirements.  So, a useful start to the
>>         conversation
>>         might be: what do you want to accomplish?
>>
>>         -- Alex
>>
>>         --
>>         Alex Balashov - Principal
>>         Evariste Systems LLC
>>         235 E Ponce de Leon Ave
>>         Suite 106
>>         Decatur, GA 30030
>>         Tel: +1-678-954-0670
>>         Fax: +1-404-961-1892
>>         Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>>
>>
>>
>>         ------------------------------
>>
>>         ______________________________**_________________
>>         sr-users mailing list
>>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-**
>> router.org <sr-users at lists.sip-router.org>>
>>
>>         http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
>>         End of sr-users Digest, Vol 86, Issue 85
>>         ******************************************
>>
>>
>>
>>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
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From abalashov at evaristesys.com  Thu Aug  2 02:15:50 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 01 Aug 2012 20:15:50 -0400
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
	<CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
Message-ID: <5019C6B6.7080906@evaristesys.com>

On 08/01/2012 08:12 PM, Ananth Kollipara wrote:

> Which module does dynamic routing?

Lots of modules do various kinds of dynamic routing: drouting, lcr, 
dialplan, etc.

But, they're rather formulaic.  If you're after something custom, just 
do it yourself using your TimesTen DB, db_unixodbc, and the sqlops 
module, which allows custom SQL queries and interactions:

    http://www.kamailio.org/docs/modules/3.3.x/modules_k/sqlops.html

And, sending 302 redirects with the desired Contact header is very easy:

    append_to_reply("Contact: <sip:...>\r\n");
    sl_send_reply("302", "Moved Temporarily");

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From infoananthk at gmail.com  Thu Aug  2 02:30:10 2012
From: infoananthk at gmail.com (Ananth Kollipara)
Date: Thu, 2 Aug 2012 06:00:10 +0530
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <5019C6B6.7080906@evaristesys.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
	<CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
	<5019C6B6.7080906@evaristesys.com>
Message-ID: <CACJ9Faup1Tu1zfHPOwgsDhkfiSu=5+m-jvujsbwMmsEhqrzUOw@mail.gmail.com>

I want the proxy to re-route the request, without informing the client.
Lets say - proxy receives message, forwards to back-end servers. If
back-end server responds with 302 response, proxy shall use the contact
header and then forwards the message to the correct backend server. Do we
have support for this?

On Thu, Aug 2, 2012 at 5:45 AM, Alex Balashov <abalashov at evaristesys.com>wrote:

> On 08/01/2012 08:12 PM, Ananth Kollipara wrote:
>
>  Which module does dynamic routing?
>>
>
> Lots of modules do various kinds of dynamic routing: drouting, lcr,
> dialplan, etc.
>
> But, they're rather formulaic.  If you're after something custom, just do
> it yourself using your TimesTen DB, db_unixodbc, and the sqlops module,
> which allows custom SQL queries and interactions:
>
>    http://www.kamailio.org/docs/**modules/3.3.x/modules_k/**sqlops.html<http://www.kamailio.org/docs/modules/3.3.x/modules_k/sqlops.html>
>
> And, sending 302 redirects with the desired Contact header is very easy:
>
>    append_to_reply("Contact: <sip:...>\r\n");
>    sl_send_reply("302", "Moved Temporarily");
>
>
> -- Alex
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
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From abalashov at evaristesys.com  Thu Aug  2 02:31:19 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 01 Aug 2012 20:31:19 -0400
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <CACJ9Faup1Tu1zfHPOwgsDhkfiSu=5+m-jvujsbwMmsEhqrzUOw@mail.gmail.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
	<CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
	<5019C6B6.7080906@evaristesys.com>
	<CACJ9Faup1Tu1zfHPOwgsDhkfiSu=5+m-jvujsbwMmsEhqrzUOw@mail.gmail.com>
Message-ID: <5019CA57.20804@evaristesys.com>

Oh, I see.  Yes, the 'uac_redirect' module can consume 302s.

On 08/01/2012 08:30 PM, Ananth Kollipara wrote:

> I want the proxy to re-route the request, without informing the client.
> Lets say - proxy receives message, forwards to back-end servers. If
> back-end server responds with 302 response, proxy shall use the contact
> header and then forwards the message to the correct backend server. Do
> we have support for this?
>
> On Thu, Aug 2, 2012 at 5:45 AM, Alex Balashov <abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>> wrote:
>
>     On 08/01/2012 08:12 PM, Ananth Kollipara wrote:
>
>         Which module does dynamic routing?
>
>
>     Lots of modules do various kinds of dynamic routing: drouting, lcr,
>     dialplan, etc.
>
>     But, they're rather formulaic.  If you're after something custom,
>     just do it yourself using your TimesTen DB, db_unixodbc, and the
>     sqlops module, which allows custom SQL queries and interactions:
>
>     http://www.kamailio.org/docs/__modules/3.3.x/modules_k/__sqlops.html
>     <http://www.kamailio.org/docs/modules/3.3.x/modules_k/sqlops.html>
>
>     And, sending 302 redirects with the desired Contact header is very easy:
>
>         append_to_reply("Contact: <sip:...>\r\n");
>         sl_send_reply("302", "Moved Temporarily");
>
>
>     -- Alex
>
>     --
>     Alex Balashov - Principal
>     Evariste Systems LLC
>     235 E Ponce de Leon Ave
>     Suite 106
>     Decatur, GA 30030
>     Tel: +1-678-954-0670
>     Fax: +1-404-961-1892
>     Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From forums at akan.net  Thu Aug  2 02:33:35 2012
From: forums at akan.net (Akan)
Date: Wed, 01 Aug 2012 19:33:35 -0500
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <50198ACE.7030909@gmail.com>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
Message-ID: <5019CADF.4070703@akan.net>

I had to upgrade to the lastest version of Postgres to get past this 
error but then ran into the problem of  "undefined symbol TLSv1_method"

Thanks

Nathaniel
On 8/1/2012 3:00 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> can you paste here the output of following command executed in the 
> db_postgres module directory:
>
> make Q=0
>
> First run 'make proper' in the same directory. I want to see the 
> compile flags and linked libs used for your system.
>
> Cheers,
> Daniel
>
> On 8/1/12 5:33 PM, Andrew O. Zhukov wrote:
>> The same trouble with:
>> completely updated Centos 5
>> the last Kamailio RPM from
>> http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/ 
>>
>> and
>> postgresql-libs.x86_64.8.1.23-5.el5_8
>>
>> several month ago I try
>> postgresql84-libs.x86_64
>> with the same result.
>>
>> It's not possible to use Kamailio RPM with postgres backend. Need to 
>> assemble it manually.
>>
>>
>> Bruno Bresciani wrote:
>> > Hi,
>> >
>> > I configure the Kamailio 3.1.2 with postgres but I cann't start. In 
>> the
>> > log file is generated the following error:
>> >
>> > //
>> >
>> > ERROR: load_module: could not open module
>> > </home2/local/kamailio/lib/kamailio/modules/db_
>> > postgres.so>: 
>> /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
>> > undefined symbol: PQdescribePrepared
>>
>> Have you built Kamailio yourself? Looks like db_postgres.so can not find
>> the postgresql libraries, or it was built with a different library 
>> version.
>>
>> Maybe ldd can give you some details:
>>
>>    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so
>>
>> regards
>> Klaus
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From miconda at gmail.com  Thu Aug  2 09:07:28 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 02 Aug 2012 09:07:28 +0200
Subject: [SR-Users] planning release of v3.3.1
In-Reply-To: <50177DC2.8080505@gmail.com>
References: <50177DC2.8080505@gmail.com>
Message-ID: <501A2730.40606@gmail.com>

Hello,

short note to remind about the release of v3.3.1 later today. Apply your 
commits to branch 3.3 before 12:00 GMT.

Cheers,
Daniel

On 7/31/12 8:40 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I am planning to package v3.3.1 out of the latest branch 3.3 this 
> Thursday, August 2. If you are aware of issues that are not listed on 
> the tracker (http://sip-router.org/tracker/), add them there to see 
> what can be sorted out before.
>
> Cheers,
> Daniel
>


-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From miconda at gmail.com  Thu Aug  2 09:10:15 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 02 Aug 2012 09:10:15 +0200
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <5019CADF.4070703@akan.net>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
	<5019CADF.4070703@akan.net>
Message-ID: <501A27D7.90107@gmail.com>

TLSV1_method is not used inside db_postgres, so it is a dependency of pg 
library. Do you have the tool pg_config installed in your system?

Cheers,
Daniel

On 8/2/12 2:33 AM, Akan wrote:
> I had to upgrade to the lastest version of Postgres to get past this 
> error but then ran into the problem of  "undefined symbol TLSv1_method"
>
> Thanks
>
> Nathaniel
> On 8/1/2012 3:00 PM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> can you paste here the output of following command executed in the 
>> db_postgres module directory:
>>
>> make Q=0
>>
>> First run 'make proper' in the same directory. I want to see the 
>> compile flags and linked libs used for your system.
>>
>> Cheers,
>> Daniel
>>
>> On 8/1/12 5:33 PM, Andrew O. Zhukov wrote:
>>> The same trouble with:
>>> completely updated Centos 5
>>> the last Kamailio RPM from
>>> http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/ 
>>>
>>> and
>>> postgresql-libs.x86_64.8.1.23-5.el5_8
>>>
>>> several month ago I try
>>> postgresql84-libs.x86_64
>>> with the same result.
>>>
>>> It's not possible to use Kamailio RPM with postgres backend. Need to 
>>> assemble it manually.
>>>
>>>
>>> Bruno Bresciani wrote:
>>> > Hi,
>>> >
>>> > I configure the Kamailio 3.1.2 with postgres but I cann't start. 
>>> In the
>>> > log file is generated the following error:
>>> >
>>> > //
>>> >
>>> > ERROR: load_module: could not open module
>>> > </home2/local/kamailio/lib/kamailio/modules/db_
>>> > postgres.so>: 
>>> /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
>>> > undefined symbol: PQdescribePrepared
>>>
>>> Have you built Kamailio yourself? Looks like db_postgres.so can not 
>>> find
>>> the postgresql libraries, or it was built with a different library 
>>> version.
>>>
>>> Maybe ldd can give you some details:
>>>
>>>    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so
>>>
>>> regards
>>> Klaus
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From infoananthk at gmail.com  Thu Aug  2 10:55:29 2012
From: infoananthk at gmail.com (Ananth Kollipara)
Date: Thu, 2 Aug 2012 14:25:29 +0530
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <5019CA57.20804@evaristesys.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
	<CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
	<5019C6B6.7080906@evaristesys.com>
	<CACJ9Faup1Tu1zfHPOwgsDhkfiSu=5+m-jvujsbwMmsEhqrzUOw@mail.gmail.com>
	<5019CA57.20804@evaristesys.com>
Message-ID: <CACJ9FatPOpty+apxxM_8a2AZxgvBCya23-jYbcqhb1asbep6Sg@mail.gmail.com>

Alex,

uac_redirect doesn't work for me. Please see the below link
http://blog.gmane.org/gmane.comp.voip.ser/month=20050201/page=10

In the above link, check the email from "Jan Janak" on Feb 3rd, with the
"call flow". I exactly require the same. Do we have some handling for this?

Regards,
Ananth





On Thu, Aug 2, 2012 at 6:01 AM, Alex Balashov <abalashov at evaristesys.com>wrote:

> Oh, I see.  Yes, the 'uac_redirect' module can consume 302s.
>
>
> On 08/01/2012 08:30 PM, Ananth Kollipara wrote:
>
>  I want the proxy to re-route the request, without informing the client.
>> Lets say - proxy receives message, forwards to back-end servers. If
>> back-end server responds with 302 response, proxy shall use the contact
>> header and then forwards the message to the correct backend server. Do
>> we have support for this?
>>
>> On Thu, Aug 2, 2012 at 5:45 AM, Alex Balashov <abalashov at evaristesys.com
>> <mailto:abalashov at evaristesys.**com <abalashov at evaristesys.com>>> wrote:
>>
>>     On 08/01/2012 08:12 PM, Ananth Kollipara wrote:
>>
>>         Which module does dynamic routing?
>>
>>
>>     Lots of modules do various kinds of dynamic routing: drouting, lcr,
>>     dialplan, etc.
>>
>>     But, they're rather formulaic.  If you're after something custom,
>>     just do it yourself using your TimesTen DB, db_unixodbc, and the
>>     sqlops module, which allows custom SQL queries and interactions:
>>
>>     http://www.kamailio.org/docs/_**_modules/3.3.x/modules_k/__**
>> sqlops.html<http://www.kamailio.org/docs/__modules/3.3.x/modules_k/__sqlops.html>
>>
>>     <http://www.kamailio.org/docs/**modules/3.3.x/modules_k/**sqlops.html<http://www.kamailio.org/docs/modules/3.3.x/modules_k/sqlops.html>
>> >
>>
>>     And, sending 302 redirects with the desired Contact header is very
>> easy:
>>
>>         append_to_reply("Contact: <sip:...>\r\n");
>>         sl_send_reply("302", "Moved Temporarily");
>>
>>
>>     -- Alex
>>
>>     --
>>     Alex Balashov - Principal
>>     Evariste Systems LLC
>>     235 E Ponce de Leon Ave
>>     Suite 106
>>     Decatur, GA 30030
>>     Tel: +1-678-954-0670
>>     Fax: +1-404-961-1892
>>     Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>>
>>
>>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
-------------- next part --------------
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From abalashov at evaristesys.com  Thu Aug  2 11:09:28 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 02 Aug 2012 05:09:28 -0400
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <CACJ9FatPOpty+apxxM_8a2AZxgvBCya23-jYbcqhb1asbep6Sg@mail.gmail.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
	<CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
	<5019C6B6.7080906@evaristesys.com>
	<CACJ9Faup1Tu1zfHPOwgsDhkfiSu=5+m-jvujsbwMmsEhqrzUOw@mail.gmail.com>
	<5019CA57.20804@evaristesys.com>
	<CACJ9FatPOpty+apxxM_8a2AZxgvBCya23-jYbcqhb1asbep6Sg@mail.gmail.com>
Message-ID: <501A43C8.1000207@evaristesys.com>

You'll have to be more specific about why it "doesn't work".

Also, keep in mind that 'uac_redirect' is just a convenience.  If it's 
not doing what you want, just build a failure_route[], capture the 302 
yourself, extract the Contact header yourself, parse the data yourself, 
and manually fork through the contacts as you please.

On 08/02/2012 04:55 AM, Ananth Kollipara wrote:

> Alex,
>
> uac_redirect doesn't work for me. Please see the below link
> http://blog.gmane.org/gmane.comp.voip.ser/month=20050201/page=10
>
> In the above link, check the email from "Jan Janak" on Feb 3rd, with the
> "call flow". I exactly require the same. Do we have some handling for this?
>
> Regards,
> Ananth
>
>
>
>
>
> On Thu, Aug 2, 2012 at 6:01 AM, Alex Balashov <abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>> wrote:
>
>     Oh, I see.  Yes, the 'uac_redirect' module can consume 302s.
>
>
>     On 08/01/2012 08:30 PM, Ananth Kollipara wrote:
>
>         I want the proxy to re-route the request, without informing the
>         client.
>         Lets say - proxy receives message, forwards to back-end servers. If
>         back-end server responds with 302 response, proxy shall use the
>         contact
>         header and then forwards the message to the correct backend
>         server. Do
>         we have support for this?
>
>         On Thu, Aug 2, 2012 at 5:45 AM, Alex Balashov
>         <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>
>         <mailto:abalashov at evaristesys.__com
>         <mailto:abalashov at evaristesys.com>>> wrote:
>
>              On 08/01/2012 08:12 PM, Ananth Kollipara wrote:
>
>                  Which module does dynamic routing?
>
>
>              Lots of modules do various kinds of dynamic routing:
>         drouting, lcr,
>              dialplan, etc.
>
>              But, they're rather formulaic.  If you're after something
>         custom,
>              just do it yourself using your TimesTen DB, db_unixodbc,
>         and the
>              sqlops module, which allows custom SQL queries and
>         interactions:
>
>         http://www.kamailio.org/docs/____modules/3.3.x/modules_k/____sqlops.html
>         <http://www.kamailio.org/docs/__modules/3.3.x/modules_k/__sqlops.html>
>
>
>         <http://www.kamailio.org/docs/__modules/3.3.x/modules_k/__sqlops.html
>         <http://www.kamailio.org/docs/modules/3.3.x/modules_k/sqlops.html>>
>
>              And, sending 302 redirects with the desired Contact header
>         is very easy:
>
>                  append_to_reply("Contact: <sip:...>\r\n");
>                  sl_send_reply("302", "Moved Temporarily");
>
>
>              -- Alex
>
>              --
>              Alex Balashov - Principal
>              Evariste Systems LLC
>              235 E Ponce de Leon Ave
>              Suite 106
>              Decatur, GA 30030
>              Tel: +1-678-954-0670
>              Fax: +1-404-961-1892
>              Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>
>
>
>     --
>     Alex Balashov - Principal
>     Evariste Systems LLC
>     235 E Ponce de Leon Ave
>     Suite 106
>     Decatur, GA 30030
>     Tel: +1-678-954-0670
>     Fax: +1-404-961-1892
>     Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From gnugk at telegroup.com.ua  Thu Aug  2 11:10:27 2012
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Thu, 02 Aug 2012 12:10:27 +0300
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <50198ACE.7030909@gmail.com>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
Message-ID: <501A4403.7060800@telegroup.com.ua>

[root at new db_postgres]# make Q=0
config.mak included
config.mak included
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
km_dbase.c -o km_dbase.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
km_db_postgres.c -o km_db_postgres.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
km_pg_con.c -o km_pg_con.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
km_res.c -o km_res.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
km_val.c -o km_val.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_cmd.c -o pg_cmd.o -MMD -MP
pg_cmd.c: In function ?get_types?:
pg_cmd.c:206: warning: implicit declaration of function ?PQdescribePrepared?
pg_cmd.c:206: warning: assignment makes pointer from integer without a cast
pg_cmd.c: In function ?pg_cmd_exec?:
pg_cmd.c:449: warning: assignment makes pointer from integer without a cast
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_con.c -o pg_con.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_fld.c -o pg_fld.o -MMD -MP
pg_fld.c: In function ?pg_resolve_param_oids?:
pg_fld.c:466: warning: implicit declaration of function ?PQnparams?
pg_fld.c:473: warning: implicit declaration of function ?PQparamtype?
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_mod.c -o pg_mod.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_oid.c -o pg_oid.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_res.c -o pg_res.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_sql.c -o pg_sql.o -MMD -MP
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron 
-Wall    -DNAME='"kamailio"' -DVERSION='"3.3.0"' -DARCH='"x86_64"' 
-DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' 
-D__CPU_x86_64 -D__OS_linux -DSER_VER=3003000 
-DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP 
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE 
-DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST 
-DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS 
-DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT 
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM 
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD 
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H 
-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL 
-DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT 
-I/usr/include -DSER_MOD_INTERFACE -DMOD_NAME='"db_postgres"' -c 
pg_uri.c -o pg_uri.o -MMD -MP
Makefile.defs defs skipped
make[1]: Entering directory `/root/kamailio-3.3.0/lib/srdb2'
make[1]: `libsrdb2.so.1.0' is up to date.
make[1]: Leaving directory `/root/kamailio-3.3.0/lib/srdb2'
Makefile.defs defs skipped
make[1]: Entering directory `/root/kamailio-3.3.0/lib/srdb1'
make[1]: `libsrdb1.so.1.0' is up to date.
make[1]: Leaving directory `/root/kamailio-3.3.0/lib/srdb1'
gcc -shared  -m64 -Wl,-O2 -Wl,-E    km_dbase.o km_db_postgres.o 
km_pg_con.o km_res.o km_val.o pg_cmd.o pg_con.o pg_fld.o pg_mod.o 
pg_oid.o pg_res.o pg_sql.o pg_uri.o  -L/usr/lib64 -lpq 
-L../../lib/srdb2/ -lsrdb2  -L../../lib/srdb1/ -lsrdb1 
-Wl,-rpath,/root/kamailio-3.3.0/lib/srdb2 
-Wl,-rpath,/root/kamailio-3.3.0/lib/srdb1 -o db_postgres.so




01.08.2012 23:00, Daniel-Constantin Mierla ?????:
> Hello,
>
> can you paste here the output of following command executed in the
> db_postgres module directory:
>
> make Q=0
>
> First run 'make proper' in the same directory. I want to see the compile
> flags and linked libs used for your system.
>
> Cheers,
> Daniel
>
> On 8/1/12 5:33 PM, Andrew O. Zhukov wrote:
>> The same trouble with:
>> completely updated Centos 5
>> the last Kamailio RPM from
>> http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/
>>
>> and
>> postgresql-libs.x86_64.8.1.23-5.el5_8
>>
>> several month ago I try
>> postgresql84-libs.x86_64
>> with the same result.
>>
>> It's not possible to use Kamailio RPM with postgres backend. Need to
>> assemble it manually.
>>
>>
>> Bruno Bresciani wrote:
>> > Hi,
>> >
>> > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the
>> > log file is generated the following error:
>> >
>> > //
>> >
>> > ERROR: load_module: could not open module
>> > </home2/local/kamailio/lib/kamailio/modules/db_
>> > postgres.so>:
>> /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
>> > undefined symbol: PQdescribePrepared
>>
>> Have you built Kamailio yourself? Looks like db_postgres.so can not find
>> the postgresql libraries, or it was built with a different library
>> version.
>>
>> Maybe ldd can give you some details:
>>
>>    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so
>>
>> regards
>> Klaus
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From gnugk at telegroup.com.ua  Thu Aug  2 11:40:27 2012
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Thu, 02 Aug 2012 12:40:27 +0300
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <50198ACE.7030909@gmail.com>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
Message-ID: <501A4B0B.2020805@telegroup.com.ua>

Daniel,

I remove RPM-s

postgresql
postgresql-devel
postgresql-libs
kamailio-postgres

install

postgresql84
postgresql84-devel
postgresql84-libs

and assemble

db_postgres using
make Q=0

in db_postgres directory.

then put
db_postgres.so to /usr/lib64/kamailio/modules directory.

Currently I's only one way to fix it.

Yum rpm installation whant postgresql-libs depedency
rpmbuild from src rpm whant postgresql-devel depedency

Thanks.


01.08.2012 23:00, Daniel-Constantin Mierla ?????:
> Hello,
>
> can you paste here the output of following command executed in the
> db_postgres module directory:
>
> make Q=0
>
> First run 'make proper' in the same directory. I want to see the compile
> flags and linked libs used for your system.
>
> Cheers,
> Daniel
>
> On 8/1/12 5:33 PM, Andrew O. Zhukov wrote:
>> The same trouble with:
>> completely updated Centos 5
>> the last Kamailio RPM from
>> http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/
>>
>> and
>> postgresql-libs.x86_64.8.1.23-5.el5_8
>>
>> several month ago I try
>> postgresql84-libs.x86_64
>> with the same result.
>>
>> It's not possible to use Kamailio RPM with postgres backend. Need to
>> assemble it manually.
>>
>>
>> Bruno Bresciani wrote:
>> > Hi,
>> >
>> > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the
>> > log file is generated the following error:
>> >
>> > //
>> >
>> > ERROR: load_module: could not open module
>> > </home2/local/kamailio/lib/kamailio/modules/db_
>> > postgres.so>:
>> /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
>> > undefined symbol: PQdescribePrepared
>>
>> Have you built Kamailio yourself? Looks like db_postgres.so can not find
>> the postgresql libraries, or it was built with a different library
>> version.
>>
>> Maybe ldd can give you some details:
>>
>>    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so
>>
>> regards
>> Klaus
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From infoananthk at gmail.com  Thu Aug  2 13:16:59 2012
From: infoananthk at gmail.com (Ananth Kollipara)
Date: Thu, 2 Aug 2012 16:46:59 +0530
Subject: [SR-Users] Regarding - Add your own module
In-Reply-To: <501A43C8.1000207@evaristesys.com>
References: <gul2qb2vat5tl7p176dprkit.1343865582901@email.android.com>
	<CACJ9FasrnKeJfxETNPRrC1EzgmkfEUT-66VKAgjQgB6WVqkcLQ@mail.gmail.com>
	<5019C59A.7000301@evaristesys.com>
	<CACJ9FauQ-FH4hq7gvoe5jya+miZr5vTDenHrP3onrk7XKG=STA@mail.gmail.com>
	<5019C6B6.7080906@evaristesys.com>
	<CACJ9Faup1Tu1zfHPOwgsDhkfiSu=5+m-jvujsbwMmsEhqrzUOw@mail.gmail.com>
	<5019CA57.20804@evaristesys.com>
	<CACJ9FatPOpty+apxxM_8a2AZxgvBCya23-jYbcqhb1asbep6Sg@mail.gmail.com>
	<501A43C8.1000207@evaristesys.com>
Message-ID: <CACJ9FaupybE1zhuat1=KyUw7sy6dcwnNazKc+zZgT0vOvTcj7Q@mail.gmail.com>

Thanks Alex. Failure_route should suffice.

On Thu, Aug 2, 2012 at 2:39 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> You'll have to be more specific about why it "doesn't work".
>
> Also, keep in mind that 'uac_redirect' is just a convenience.  If it's not
> doing what you want, just build a failure_route[], capture the 302
> yourself, extract the Contact header yourself, parse the data yourself, and
> manually fork through the contacts as you please.
>
>
> On 08/02/2012 04:55 AM, Ananth Kollipara wrote:
>
>  Alex,
>>
>> uac_redirect doesn't work for me. Please see the below link
>> http://blog.gmane.org/gmane.**comp.voip.ser/month=20050201/**page=10<http://blog.gmane.org/gmane.comp.voip.ser/month=20050201/page=10>
>>
>> In the above link, check the email from "Jan Janak" on Feb 3rd, with the
>> "call flow". I exactly require the same. Do we have some handling for
>> this?
>>
>> Regards,
>> Ananth
>>
>>
>>
>>
>>
>> On Thu, Aug 2, 2012 at 6:01 AM, Alex Balashov <abalashov at evaristesys.com
>> <mailto:abalashov at evaristesys.**com <abalashov at evaristesys.com>>> wrote:
>>
>>     Oh, I see.  Yes, the 'uac_redirect' module can consume 302s.
>>
>>
>>     On 08/01/2012 08:30 PM, Ananth Kollipara wrote:
>>
>>         I want the proxy to re-route the request, without informing the
>>         client.
>>         Lets say - proxy receives message, forwards to back-end servers.
>> If
>>         back-end server responds with 302 response, proxy shall use the
>>         contact
>>         header and then forwards the message to the correct backend
>>         server. Do
>>         we have support for this?
>>
>>         On Thu, Aug 2, 2012 at 5:45 AM, Alex Balashov
>>         <abalashov at evaristesys.com <mailto:abalashov at evaristesys.**com<abalashov at evaristesys.com>
>> >
>>         <mailto:abalashov at evaristesys.**__com
>>
>>         <mailto:abalashov at evaristesys.**com <abalashov at evaristesys.com>>>>
>> wrote:
>>
>>              On 08/01/2012 08:12 PM, Ananth Kollipara wrote:
>>
>>                  Which module does dynamic routing?
>>
>>
>>              Lots of modules do various kinds of dynamic routing:
>>         drouting, lcr,
>>              dialplan, etc.
>>
>>              But, they're rather formulaic.  If you're after something
>>         custom,
>>              just do it yourself using your TimesTen DB, db_unixodbc,
>>         and the
>>              sqlops module, which allows custom SQL queries and
>>         interactions:
>>
>>         http://www.kamailio.org/docs/_**___modules/3.3.x/modules_k/___**
>> _sqlops.html<http://www.kamailio.org/docs/____modules/3.3.x/modules_k/____sqlops.html>
>>         <http://www.kamailio.org/docs/**__modules/3.3.x/modules_k/__**
>> sqlops.html<http://www.kamailio.org/docs/__modules/3.3.x/modules_k/__sqlops.html>
>> >
>>
>>
>>
>>         <http://www.kamailio.org/docs/**__modules/3.3.x/modules_k/__**
>> sqlops.html<http://www.kamailio.org/docs/__modules/3.3.x/modules_k/__sqlops.html>
>>         <http://www.kamailio.org/docs/**modules/3.3.x/modules_k/**
>> sqlops.html<http://www.kamailio.org/docs/modules/3.3.x/modules_k/sqlops.html>
>> >>
>>
>>              And, sending 302 redirects with the desired Contact header
>>         is very easy:
>>
>>                  append_to_reply("Contact: <sip:...>\r\n");
>>                  sl_send_reply("302", "Moved Temporarily");
>>
>>
>>              -- Alex
>>
>>              --
>>              Alex Balashov - Principal
>>              Evariste Systems LLC
>>              235 E Ponce de Leon Ave
>>              Suite 106
>>              Decatur, GA 30030
>>              Tel: +1-678-954-0670
>>              Fax: +1-404-961-1892
>>              Web: http://www.evaristesys.com/,
>> http://www.alexbalashov.com/
>>
>>
>>
>>
>>     --
>>     Alex Balashov - Principal
>>     Evariste Systems LLC
>>     235 E Ponce de Leon Ave
>>     Suite 106
>>     Decatur, GA 30030
>>     Tel: +1-678-954-0670
>>     Fax: +1-404-961-1892
>>     Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>>
>>
>>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
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From peter.dunkley at crocodile-rcs.com  Thu Aug  2 16:15:38 2012
From: peter.dunkley at crocodile-rcs.com (Peter Dunkley)
Date: Thu, 02 Aug 2012 15:15:38 +0100
Subject: [SR-Users] [sr-dev]  Outbound and Registrar. No binding update
In-Reply-To: <CABw3bnPQBapqcTj7oZNdUEo4PPwLQAoRrkyQE0r9Rd9PiZKChw@mail.gmail.com>
References: <CABw3bnPLyckUJ=UXAJLpDuvdEesmv_Yr3Th45FQDsOR9zxbQTg@mail.gmail.com>
	<CALiegf=4kAfdJTdm0foF2rnbTM_War1+2+8KZRLzUk0RiNUGzg@mail.gmail.com>
	<50127C9D.7070908@gmail.com>
	<CABw3bnMjjeLO8zg3cKLm4ths1t1VFpEFY6A6jpObu1COKzHbrw@mail.gmail.com>
	<50165CC2.7050400@gmail.com> <50167004.7060301@gmail.com>
	<CALiegf=3nGPYedATnWhqRtwsZyGQZn0_O=7n78QFwv6aBoDu1A@mail.gmail.com>
	<5016BE2F.6010300@gmail.com>
	<CALiegfm4RzkYXLamBLE3oZc=kT2C1Ki0wcEkQu7T9NVJEgByyw@mail.gmail.com>
	<50177E6F.4030003@gmail.com>
	<CABw3bnPQBapqcTj7oZNdUEo4PPwLQAoRrkyQE0r9Rd9PiZKChw@mail.gmail.com>
Message-ID: <1343916938.7084.7.camel@pd-notebook-linux.croc.internal>

Hello,

What I am trying to work out is whether Kamailio now supports enough of
Outbound and GRUU that the contact aliasing I am using with WebSockets
is no longer necessary?

Also, is the current support complete enough that there are no issues
around WebSocket connections being broken and the client re-registering
(effectively replacing the existing contact binding)?

I believe for WebSockets the situation is exactly the same as you would
have for a client connecting over TCP from behind a NAT.

Regards,

Peter

On Tue, 2012-07-31 at 13:59 +0200, Jos? Luis Mill?n wrote:

> Hi,
> 
> 
> 
> Outbound provides the UAC a way to update a binding even if it
> reboots. For that, a unique and permanent value of instance-id is
> used, which in conjunction with the AoR and reg-id determines the
> binding to the UAC.
> 
> 
> Having said this, I guess that the CSeq comparison between the one in
> the Register request and the one in the binding does not apply in this
> scenario since it is not guaranteed that a UAC saves the CSeq value of
> the registration among reboots.
> 
> 
> Regards.
> 
> 
> 2012/7/31 Daniel-Constantin Mierla <miconda at gmail.com>
> 
>         Hello,
>         
>         
>         
>         On 7/30/12 7:23 PM, I?aki Baz Castillo wrote:
>         
>                 2012/7/30 Daniel-Constantin Mierla
>                 <miconda at gmail.com>:
>                 
>                         quick question to double check if what I
>                         understood when I read the specs
>                         was ok -- in gruu/ob, it does not matter
>                         anymore the callid/cseq
>                         combination, or there should still be some
>                         checks related to it?
>                 
>                 In fact that depends on Outbound instead of GRUU, and
>                 not, when using
>                 Outbound the registrar does NOT check the Call-ID and
>                 CSeq of the
>                 REGISTER (and using GRUU means that Outbound is also
>                 used, so the
>                 Contact has a +sip.instance and reg-id params which
>                 are inspected by
>                 the registrar to create/update/delete a binding).
>         
>         but what about same callid with lower cseq, combined with same
>         sip instance and reg-id?
>         
>         
>         Cheers,
>         Daniel
>         
>         
>         _______________________________________________
>         sr-dev mailing list
>         sr-dev at lists.sip-router.org
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
> 
> 
> 

-- 
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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From ibc at aliax.net  Thu Aug  2 16:21:31 2012
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 2 Aug 2012 16:21:31 +0200
Subject: [SR-Users] [sr-dev] Outbound and Registrar. No binding update
In-Reply-To: <5016C1D3.6090105@gmail.com>
References: <CABw3bnPLyckUJ=UXAJLpDuvdEesmv_Yr3Th45FQDsOR9zxbQTg@mail.gmail.com>
	<CALiegf=4kAfdJTdm0foF2rnbTM_War1+2+8KZRLzUk0RiNUGzg@mail.gmail.com>
	<50127C9D.7070908@gmail.com>
	<CABw3bnMjjeLO8zg3cKLm4ths1t1VFpEFY6A6jpObu1COKzHbrw@mail.gmail.com>
	<50165CC2.7050400@gmail.com> <50167004.7060301@gmail.com>
	<CALiegf=3nGPYedATnWhqRtwsZyGQZn0_O=7n78QFwv6aBoDu1A@mail.gmail.com>
	<CABw3bnM2SkoBegRzTO6ww4cknixSHa58v2MEOD7fBZtw_6tSeA@mail.gmail.com>
	<1343664023.16452.4.camel@pd-notebook-linux.croc.internal>
	<CALiegf=rFxizowjWL1OeiKb2tjdqubcL5KOx+4saAXYY0hLbzw@mail.gmail.com>
	<422044d2e497bc7a919fc880df7d1b57.squirrel@crocodile-rcs.com>
	<5016C1D3.6090105@gmail.com>
Message-ID: <CALiegfnYEpQHGn3tnry0CPA2zsDmW2pj1s-WUyr88FjT+r+XFg@mail.gmail.com>

2012/7/30 Daniel-Constantin Mierla <miconda at gmail.com>:
> There is one situation that will not work even with gruu/ob -- in sip a
> phone can call without registering. A gruu contact is obtained via
> registration and then used for next requests by UA itself. By calling
> without registering, there is no gruu contact for it, so adding the src ip
> and port as alias parameter is still needed. I don't remember I have seen
> any rfc making registration mandatory before calling.

Hi Daniel, RFC 5626 (Outbound) assumes that the UA registers after
connecting to the (Outbound Edge) Proxy.


> In other words, just to summarize the gruu versus contact aliasing.

I don't think this iw "gruu versus contact aliasing" but "outbound
versus contact aliasing".
Outbound means that the proxy inserts a flow token in the Record-Route
username and then, when a request with Route header arrives to the
proxy, its Route username is inspected and the associated connection
retrieved for routing the request without inspecting the RURI (unless
it's a GRUU RURI).


Regards.


-- 
I?aki Baz Castillo
<ibc at aliax.net>


From ibc at aliax.net  Thu Aug  2 16:23:19 2012
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 2 Aug 2012 16:23:19 +0200
Subject: [SR-Users] Outbound and Registrar. No binding update
In-Reply-To: <CABw3bnPQBapqcTj7oZNdUEo4PPwLQAoRrkyQE0r9Rd9PiZKChw@mail.gmail.com>
References: <CABw3bnPLyckUJ=UXAJLpDuvdEesmv_Yr3Th45FQDsOR9zxbQTg@mail.gmail.com>
	<CALiegf=4kAfdJTdm0foF2rnbTM_War1+2+8KZRLzUk0RiNUGzg@mail.gmail.com>
	<50127C9D.7070908@gmail.com>
	<CABw3bnMjjeLO8zg3cKLm4ths1t1VFpEFY6A6jpObu1COKzHbrw@mail.gmail.com>
	<50165CC2.7050400@gmail.com> <50167004.7060301@gmail.com>
	<CALiegf=3nGPYedATnWhqRtwsZyGQZn0_O=7n78QFwv6aBoDu1A@mail.gmail.com>
	<5016BE2F.6010300@gmail.com>
	<CALiegfm4RzkYXLamBLE3oZc=kT2C1Ki0wcEkQu7T9NVJEgByyw@mail.gmail.com>
	<50177E6F.4030003@gmail.com>
	<CABw3bnPQBapqcTj7oZNdUEo4PPwLQAoRrkyQE0r9Rd9PiZKChw@mail.gmail.com>
Message-ID: <CALiegfkFq=LQf6n2N+kCJ3aJvc=gauy7Bj8m5oUtzdq+YSpkdQ@mail.gmail.com>

2012/7/31 Jos? Luis Mill?n <jmillan at aliax.net>:
> Outbound provides the UAC a way to update a binding even if it reboots. For
> that, a unique and permanent value of instance-id is used, which in
> conjunction with the AoR and reg-id determines the binding to the UAC.
>
> Having said this, I guess that the CSeq comparison between the one in the
> Register request and the one in the binding does not apply in this scenario
> since it is not guaranteed that a UAC saves the CSeq value of the
> registration among reboots.

This is correct. According to RFC 5626, when the REGISTER includes a
Contact with +sip.instance and reg-id params the registrar MUST NOT
check the call-id and cseq of the request, but just the +sip.instance
and reg-id params (and the registering AoR in the To header of
course).


-- 
I?aki Baz Castillo
<ibc at aliax.net>


From ibc at aliax.net  Thu Aug  2 16:23:53 2012
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 2 Aug 2012 16:23:53 +0200
Subject: [SR-Users] [sr-dev]  Outbound and Registrar. No binding update
In-Reply-To: <1343916938.7084.7.camel@pd-notebook-linux.croc.internal>
References: <CABw3bnPLyckUJ=UXAJLpDuvdEesmv_Yr3Th45FQDsOR9zxbQTg@mail.gmail.com>
	<CALiegf=4kAfdJTdm0foF2rnbTM_War1+2+8KZRLzUk0RiNUGzg@mail.gmail.com>
	<50127C9D.7070908@gmail.com>
	<CABw3bnMjjeLO8zg3cKLm4ths1t1VFpEFY6A6jpObu1COKzHbrw@mail.gmail.com>
	<50165CC2.7050400@gmail.com> <50167004.7060301@gmail.com>
	<CALiegf=3nGPYedATnWhqRtwsZyGQZn0_O=7n78QFwv6aBoDu1A@mail.gmail.com>
	<5016BE2F.6010300@gmail.com>
	<CALiegfm4RzkYXLamBLE3oZc=kT2C1Ki0wcEkQu7T9NVJEgByyw@mail.gmail.com>
	<50177E6F.4030003@gmail.com>
	<CABw3bnPQBapqcTj7oZNdUEo4PPwLQAoRrkyQE0r9Rd9PiZKChw@mail.gmail.com>
	<1343916938.7084.7.camel@pd-notebook-linux.croc.internal>
Message-ID: <CALiegf=g6_RqC1y4wMJyzQRJypLEKZZWCqNkB3MdSUL9kznwqw@mail.gmail.com>

2012/8/2 Peter Dunkley <peter.dunkley at crocodile-rcs.com>

> I believe for WebSockets the situation is exactly the same as you would
> have for a client connecting over TCP from behind a NAT.


It should be exactly the same situation.

-- 
I?aki Baz Castillo
<ibc at aliax.net>
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From miconda at gmail.com  Thu Aug  2 16:32:45 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 02 Aug 2012 16:32:45 +0200
Subject: [SR-Users] Kamailio v3.3.1 Released
Message-ID: <501A8F8D.5080905@gmail.com>

Hello,

Kamailio SIP Server v3.3.1 stable release is out.

This is a maintenance release of the latest stable branch, 3.3, that 
includes fixes since release of v3.3.0. There is no change to database 
schema or configuration language structure that you have to do on 
installations of v3.3.0. Deployments running previous v3.x.x versions 
are strongly recommended to be upgraded to v3.3.1.

For more details about version 3.3.1 (including links and hints to 
download the tarball or from GIT repository), visit:

   * http://www.kamailio.org/w/2012/08/kamailio-v3-3-1-released/

RPM, Debian/Ubuntu packages will be available soon as well.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From universe at truemetal.org  Thu Aug  2 18:40:58 2012
From: universe at truemetal.org (Markus)
Date: Thu, 02 Aug 2012 18:40:58 +0200
Subject: [SR-Users] Getting started with Kamailio
Message-ID: <501AAD9A.3030406@truemetal.org>

Hello! I'm new to this list and Kamailio.

First, thanks to all the programmers that developed this software.

I'd like to ask several questions on how to get started with Kamailio 
and the basic understanding of it. I'm trying to figure out Kamailio 
only since yesterday. Right now I have a single Asterisk server that 
handles signaling and media. For accounting and user administration I'm 
using a2billing with MySQL on it. For redundancy and scalability I would 
like to create the following setup:

France:
BGP anycast 195.5.5.0/24
2 nodes with Kamailio (IP 195.5.5.5) active + standby
2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster.

Spain:
BGP anycast 195.5.5.0/24
2 nodes with Kamailio (IP 195.5.5.5) active + standby
2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster.

(a2billing MySQL DB will be always synchronous over all nodes in Spain 
and France)

This will give me:

- Network redundancy (data center burns down in France, I stay online in 
Spain). This will be handled by BGP anycast.

- Signaling redundancy (Kamailio active + standby). This will be handled 
by Linux-HA or something like that.

- Media redundancy (2x Asterisk per country). This will be handled by 
Kamailio dispatcher.

- Application/database redundancy (2x a2billing per country + 
synchronous MySQL DB everyhwere). This will be handled by DRBD and MySQL.

= High redundancy, I can sleep and it can scale.

What I would like to achieve besides the above:

- Give only a single IP address to all customers and termination 
providers (the same IP address), 195.5.5.5.

I'm a SIP noob, so I have to ask:

- How do I do the Kamailio part? ;-) ... I have seen this how-to for 
Kamailio/Asterisk realtime: 
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
But it feels like "overkill". If I would not have any users that 
actually REGISTER (e.g. in a pure wholesale termination environment), I 
would not need Kamailio/Asterisk realtime integration. Correct?

- Is there no way around changing the Asterisk side (activating 
realtime, new MySQL DB) when I have users that do need to REGISTER? If I 
would not be using a2billing I could probably handle all the 
registrations in Kamailio only?

- Since Kamailio and Asterisk will not be on the same box, what is the 
recommended way for Kamailio securely communicating with the MySQL 
database on the Asterisk server? Does Kamailio support SSL with MySQL?

- If I use RTPproxy on the Kamailio server, every customer and 
termination provider would connect to 1 single IP address, because both 
media and signaling comes from that IP. Correct?

- If I don't use RTPproxy, and have canreinvite=yes on my Asterisk 
servers, customers would get the media, when placing PSTN calls, 
directly from my termination providers (I would like to avoid that). 
Correct?

- If I don't use RTPproxy, and have canreinvite=no on my Asterisk 
servers, customers will get the media directly from my Asterisk servers, 
but termination providers that filter based on IP addresses they would 
have to allow all Asterisk IP addresses in their filters (same for 
customers, actually). Correct?

Right now I didn't have to worry about such things because media and 
signaling were handled by a single Asterisk box with canreinvite=no 
everywhere.

- Last question for now: why does it seem like important developers of 
SIP software such as Kamailio and yate are originating from Romania and 
are female? Just a coincidence? :-)

Regards
Markus




From abalashov at evaristesys.com  Fri Aug  3 00:11:06 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 02 Aug 2012 18:11:06 -0400
Subject: [SR-Users] Getting started with Kamailio
In-Reply-To: <501AAD9A.3030406@truemetal.org>
References: <501AAD9A.3030406@truemetal.org>
Message-ID: <501AFAFA.70405@evaristesys.com>

Hi Markus!

On 08/02/2012 12:40 PM, Markus wrote:

> Hello! I'm new to this list and Kamailio.

Welcome!

> - Give only a single IP address to all customers and termination
> providers (the same IP address), 195.5.5.5.

That would be difficult, unless you really have your anycast ducks in a 
row.  A more common way to approach this problem is SRV records, though 
not all SIP endpoints out there implement SRV correctly or make use of 
it in the manner contemplated by a redundancy-oriented use of them.

> I'm a SIP noob, so I have to ask:
>
> - How do I do the Kamailio part? ;-) ... I have seen this how-to for
> Kamailio/Asterisk realtime:
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
> But it feels like "overkill". If I would not have any users that
> actually REGISTER (e.g. in a pure wholesale termination environment), I
> would not need Kamailio/Asterisk realtime integration. Correct?

Correct.  The reason this tutorial exists it to try to create an 
approachable use-case and gentle introduction for those coming off of an 
Asterisk background, as you are.  The problem, of course, is that it's 
just a procedure, not a methodological or best-practical recommendation 
as to the propriety of such a setup for what you are doing.

You really don't want to use Asterisk's built-in registrar if you want 
to get the performance/density/scalability benefits of Kamailio.  If 
you're only doing IP-trust relationships, as is commonly the case in 
wholesale termination, then it's even simpler.

> - Is there no way around changing the Asterisk side (activating
> realtime, new MySQL DB) when I have users that do need to REGISTER? If I
> would not be using a2billing I could probably handle all the
> registrations in Kamailio only?

You can proxy registrations and relay them to Kamailio, but the proper 
way to do that is with the Path extension, which Asterisk does not, at 
the present time, support.

You can get around that by encoding client-side contact info in the 
Contact header, though it's not a SIP-pretty approach.

Generally speaking, though, you really want to go with a centralised 
registrar, if you're going to have to have registrations.

I don't know much about a2billing, but one problem you're going to face 
if you're going to put a dispatcher-based load balancer in front of it 
is how to get it to identify the provenance of a SIP request with an 
internal client/peer/entity.  It can't be done by source IP, since the 
source IP will always be the proxy.  It might be possible from the From 
URI or the like.  Or, you might have to add a custom header (append_hf() 
in Kamailio) hinting to the dial plan who the customer is, and fish it 
out in the dial plan (${SIP_HEADER(...)}) and pass it to a2billing. 
That would be the more common approach.

> - Since Kamailio and Asterisk will not be on the same box, what is the
> recommended way for Kamailio securely communicating with the MySQL
> database on the Asterisk server? Does Kamailio support SSL with MySQL?

Isn't that transparently implemented in libmysqlclient?

> - If I use RTPproxy on the Kamailio server, every customer and
> termination provider would connect to 1 single IP address, because both
> media and signaling comes from that IP. Correct?

They would "connect" to your signaling IP, regardless, from a 
configuration perspective.  The location of the rtpproxy just governs 
the media endpoints.

> - If I don't use RTPproxy, and have canreinvite=yes on my Asterisk
> servers, customers would get the media, when placing PSTN calls,
> directly from my termination providers (I would like to avoid that).
> Correct?

In theory, yes.  In practice, firewall restrictions on the customer or 
provider side, or other network/transport-layer reachability issues, 
could preclude that.

> - If I don't use RTPproxy, and have canreinvite=no on my Asterisk
> servers, customers will get the media directly from my Asterisk servers,
> but termination providers that filter based on IP addresses they would
> have to allow all Asterisk IP addresses in their filters (same for
> customers, actually). Correct?

Correct.

> - Last question for now: why does it seem like important developers
> of SIP software such as Kamailio and yate are originating from
> Romania and are female? Just a coincidence? :-)

In the case of Kamailio, I would say most of the developers are male, 
but it does have a non-trivial contingent of significant female 
contributors, definitely in higher proportion compared to the median 
open-source project.  In the case of Yate, that's just coincidence.

Romania is relatively famous for having a higher proportion of female 
programmers.  It has excellent foundational technical education and it 
is an established career path for a lot of women there.

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From varsha.venkataramani at gmail.com  Fri Aug  3 00:53:21 2012
From: varsha.venkataramani at gmail.com (Varsha Venkatraramani)
Date: Thu, 2 Aug 2012 15:53:21 -0700
Subject: [SR-Users] Question on ACK behavior
In-Reply-To: <3C3FF486AAEE0344B1943999088DE5AA0AC63212A2@EXCHANGE.liveops.com>
References: <3C3FF486AAEE0344B1943999088DE5AA0AC63212A2@EXCHANGE.liveops.com>
Message-ID: <CABP4ryOg0=PBNxoRnSg6u7pFDgQSKbxqsQtzN-y2fd7uUc87zQ@mail.gmail.com>

** **

*
*

** **

Hi,

** **

I have a question regarding the below ACK response Kamilaio receives from a
carrier. Can  someone please help me understand why the ?*Route:
sip:callmanager at 192.168.160.43:5060> *sent from Kamailio is missing a ?<?
as shown in the captures below? Our Internal proxy is treating that as a
malformed header and dropping the packet.

** **

** **

*ACK from CARRIER*

* *

U 2012/08/01 18:32:52.219852 4.55.18.227:5060 -> 192.168.160.47:5060****

*ACK sip:callmanager at 192.168.160.43:5060 SIP/2.0.*

Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.****

From: sip:+14088442721 at 4.55.18.227:5060;tag=gK043001e3.****

To: sip:+19728931740 at 192.168.160.47:5060;tag=b307370c678f3b44.****

Call-ID: 295226_50030734 at 4.55.18.227.****

CSeq: 18079 ACK.****

Max-Forwards: 70.****

Route: <sip:192.168.160.47:5060;lr=on>.****

*Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982 at 192.168.160.44:5060>.*

Content-Length: 0.****

.****

*ACK FORWARDED TO SIP PROXY*

** **

U 2012/08/01 18:32:52.220612 192.168.160.47:5060 -> 192.168.160.44:5060****

*ACK sip:2c6c6d1ab58c623912f6b8a6ee526982 at 192.168.160.44:5060 SIP/2.0.*

Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.****

Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.****

From: sip:+14088442721 at 4.55.18.227:5060;tag=gK043001e3.****

To: sip:+19728931740 at 192.168.160.47:5060;tag=b307370c678f3b44.****

Call-ID: 295226_50030734 at 4.55.18.227.****

CSeq: 18079 ACK.****

Max-Forwards: 69.****

Content-Length: 0.****

*Route: sip:callmanager at 192.168.160.43:5060>.*

I have attached the config file for you reference. Kamailio version is 3.2.3
****

** **

Thank You****

Varsha****
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From varsha.venkataramani at gmail.com  Fri Aug  3 01:43:16 2012
From: varsha.venkataramani at gmail.com (Varsha Venkatraramani)
Date: Thu, 2 Aug 2012 16:43:16 -0700
Subject: [SR-Users] Question on ACK behavior
In-Reply-To: <CABP4ryOg0=PBNxoRnSg6u7pFDgQSKbxqsQtzN-y2fd7uUc87zQ@mail.gmail.com>
References: <3C3FF486AAEE0344B1943999088DE5AA0AC63212A2@EXCHANGE.liveops.com>
	<CABP4ryOg0=PBNxoRnSg6u7pFDgQSKbxqsQtzN-y2fd7uUc87zQ@mail.gmail.com>
Message-ID: <CABP4ryPryWOQ05RkVjoVzeGPE72p4Qr3wPgPF9o=ERiZaOC9EQ@mail.gmail.com>

> ** **
>
> *
> *
>
> ** **
>
> Hi,
>
> ** **
>
> I have a question regarding the below ACK response Kamilaio receives from
> a carrier. Can  someone please help me understand why the ?*Route:
> sip:callmanager at 192.168.160.43:5060> *sent from Kamailio is missing a ?<?
> as shown in the captures below? Our Internal proxy is treating that as a
> malformed header and dropping the packet.
>
> ** **
>
> ** **
>
> *ACK from CARRIER*
>
> * *
>
> U 2012/08/01 18:32:52.219852 4.55.18.227:5060 -> 192.168.160.47:5060****
>
> *ACK sip:callmanager at 192.168.160.43:5060 SIP/2.0.*
>
> Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.****
>
> From: sip:+14088442721 at 4.55.18.227:5060;tag=gK043001e3.****
>
> To: sip:+19728931740 at 192.168.160.47:5060;tag=b307370c678f3b44.****
>
> Call-ID: 295226_50030734 at 4.55.18.227.****
>
> CSeq: 18079 ACK.****
>
> Max-Forwards: 70.****
>
> Route: <sip:192.168.160.47:5060;lr=on>.****
>
> *Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982 at 192.168.160.44:5060>.*
>
> Content-Length: 0.****
>
> .****
>
> *ACK FORWARDED TO SIP PROXY*
>
> ** **
>
> U 2012/08/01 18:32:52.220612 192.168.160.47:5060 -> 192.168.160.44:5060***
> *
>
> *ACK sip:2c6c6d1ab58c623912f6b8a6ee526982 at 192.168.160.44:5060 SIP/2.0.*
>
> Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.****
>
> Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.****
>
> From: sip:+14088442721 at 4.55.18.227:5060;tag=gK043001e3.****
>
> To: sip:+19728931740 at 192.168.160.47:5060;tag=b307370c678f3b44.****
>
> Call-ID: 295226_50030734 at 4.55.18.227.****
>
> CSeq: 18079 ACK.****
>
> Max-Forwards: 69.****
>
> Content-Length: 0.****
>
> *Route: sip:callmanager at 192.168.160.43:5060>.*
>
> I have attached the config file for you reference. Kamailio version is
> 3.2.3****
>
> ** **
>
> Thank You for your help/input in advance
>
> Varsha****
>
>
>
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From keeling at akan-tech.com  Fri Aug  3 05:42:57 2012
From: keeling at akan-tech.com (Nathaniel L Keeling)
Date: Thu, 02 Aug 2012 22:42:57 -0500
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <501A27D7.90107@gmail.com>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
	<5019CADF.4070703@akan.net> <501A27D7.90107@gmail.com>
Message-ID: <501B48C1.5020005@akan-tech.com>

Yes but it was for an older version that was uninstalled. I changed the 
path to pull in the current version and still received the same error. I 
checked the version for openssl and the same thing was happening. I 
changed the path again and recompiled kamailio and still no change. I 
noticed when compiling kamailio, the tls.so module was complaining that 
the version of openssl was less then 1.0. The version that I installed 
is 1.0.1c.

Thanks

Nathaniel

On 8/2/2012 2:10 AM, Daniel-Constantin Mierla wrote:
> TLSV1_method is not used inside db_postgres, so it is a dependency of 
> pg library. Do you have the tool pg_config installed in your system?
>
> Cheers,
> Daniel
>
> On 8/2/12 2:33 AM, Akan wrote:
>> I had to upgrade to the lastest version of Postgres to get past this 
>> error but then ran into the problem of  "undefined symbol TLSv1_method"
>>
>> Thanks
>>
>> Nathaniel
>> On 8/1/2012 3:00 PM, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> can you paste here the output of following command executed in the 
>>> db_postgres module directory:
>>>
>>> make Q=0
>>>
>>> First run 'make proper' in the same directory. I want to see the 
>>> compile flags and linked libs used for your system.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 8/1/12 5:33 PM, Andrew O. Zhukov wrote:
>>>> The same trouble with:
>>>> completely updated Centos 5
>>>> the last Kamailio RPM from
>>>> http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_RHEL-5/ 
>>>>
>>>> and
>>>> postgresql-libs.x86_64.8.1.23-5.el5_8
>>>>
>>>> several month ago I try
>>>> postgresql84-libs.x86_64
>>>> with the same result.
>>>>
>>>> It's not possible to use Kamailio RPM with postgres backend. Need 
>>>> to assemble it manually.
>>>>
>>>>
>>>> Bruno Bresciani wrote:
>>>> > Hi,
>>>> >
>>>> > I configure the Kamailio 3.1.2 with postgres but I cann't start. 
>>>> In the
>>>> > log file is generated the following error:
>>>> >
>>>> > //
>>>> >
>>>> > ERROR: load_module: could not open module
>>>> > </home2/local/kamailio/lib/kamailio/modules/db_
>>>> > postgres.so>: 
>>>> /home2/local/kamailio/lib/kamailio/modules/db_postgres.so:
>>>> > undefined symbol: PQdescribePrepared
>>>>
>>>> Have you built Kamailio yourself? Looks like db_postgres.so can not 
>>>> find
>>>> the postgresql libraries, or it was built with a different library 
>>>> version.
>>>>
>>>> Maybe ldd can give you some details:
>>>>
>>>>    ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so
>>>>
>>>> regards
>>>> Klaus
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>>> list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From andre at rodnoe.tv  Thu Aug  2 18:24:54 2012
From: andre at rodnoe.tv (andre)
Date: Thu, 02 Aug 2012 19:24:54 +0300
Subject: [SR-Users] centos (fedora) service startup file
Message-ID: <501AA9D6.4040500@rodnoe.tv>

Somwhere losted since openser 1.0 :)


Please fix in:

start() {
         echo -n $"Starting $prog: "
.....
    [ $RETVAL = 0 ] && touch something
to
   [ $RETVAL = 0 ] && touch someting && success

I mean call function
success
in the end of the expression.

Thanks in advance


From andre at rodnoe.tv  Thu Aug  2 20:05:16 2012
From: andre at rodnoe.tv (andre)
Date: Thu, 02 Aug 2012 21:05:16 +0300
Subject: [SR-Users] centos (fedora) service startup file
Message-ID: <501AC15C.8030801@rodnoe.tv>

Somwhere losted since openser 1.0

file kamailio.init

Please fix in function start:

start() {
         echo -n $"Starting $prog: "
.....
    [ $RETVAL = 0 ] && touch something
to
   [ $RETVAL = 0 ] && touch someting && success

I mean call function
success
in the end of the expression.

Thanks in advance.


From miconda at gmail.com  Fri Aug  3 09:33:03 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Aug 2012 09:33:03 +0200
Subject: [SR-Users] Question on ACK behavior
In-Reply-To: <CABP4ryPryWOQ05RkVjoVzeGPE72p4Qr3wPgPF9o=ERiZaOC9EQ@mail.gmail.com>
References: <3C3FF486AAEE0344B1943999088DE5AA0AC63212A2@EXCHANGE.liveops.com>
	<CABP4ryOg0=PBNxoRnSg6u7pFDgQSKbxqsQtzN-y2fd7uUc87zQ@mail.gmail.com>
	<CABP4ryPryWOQ05RkVjoVzeGPE72p4Qr3wPgPF9o=ERiZaOC9EQ@mail.gmail.com>
Message-ID: <501B7EAF.2010405@gmail.com>

Hello,

On 8/3/12 1:43 AM, Varsha Venkatraramani wrote:
>
>     I have a question regarding the below ACK response Kamilaio
>     receives from a carrier.Can  someone please help me understand why
>     the "*Route: sip:callmanager at 192.168.160.43:5060> *sent from
>     Kamailio is missing a "<" as shown in the captures below? Our
>     Internal proxy is treating that as a malformed header and dropping
>     the packet.
>
>     *ACK from CARRIER*
>
>     U 2012/08/01 18:32:52.219852 4.55.18.227:5060
>     <http://4.55.18.227:5060> -> 192.168.160.47:5060
>     <http://192.168.160.47:5060>
>
>     *ACK sip:callmanager at 192.168.160.43:5060 SIP/2.0.*
>
>     Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.
>
>     From: sip:+14088442721 at 4.55.18.227:5060;tag=gK043001e3.
>
>     To: sip:+19728931740 at 192.168.160.47:5060;tag=b307370c678f3b44.
>
>     Call-ID: 295226_50030734 at 4.55.18.227
>     <mailto:295226_50030734 at 4.55.18.227>.
>
>     CSeq: 18079 ACK.
>
>     Max-Forwards: 70.
>
>     Route: <sip:192.168.160.47:5060;lr=on>.
>
>     *Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982 at 192.168.160.44:5060>.*
>
>     Content-Length: 0.
>
>     .
>
>     *ACK FORWARDED TO SIP PROXY*
>
>     U 2012/08/01 18:32:52.220612 192.168.160.47:5060
>     <http://192.168.160.47:5060> -> 192.168.160.44:5060
>     <http://192.168.160.44:5060>
>
>     *ACK sip:2c6c6d1ab58c623912f6b8a6ee526982 at 192.168.160.44:5060
>     SIP/2.0.*
>
>     Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.
>
>     Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.
>
>     From: sip:+14088442721 at 4.55.18.227:5060;tag=gK043001e3.
>
>     To: sip:+19728931740 at 192.168.160.47:5060;tag=b307370c678f3b44.
>
>     Call-ID: 295226_50030734 at 4.55.18.227
>     <mailto:295226_50030734 at 4.55.18.227>.
>
>     CSeq: 18079 ACK.
>
>     Max-Forwards: 69.
>
>     Content-Length: 0.
>
>     *Route: sip:callmanager at 192.168.160.43:5060>.*
>
>     I have attached the config file for you reference. Kamailio
>     version is 3.2.3
>
the next proxy does strict routing, because the Route header has not 
'lr' parameter. Based on SIP specs, Kamailio has to take that route and 
set it as r-uri and the r-uri has to be added as last route, so next 
proxy will be able to do forwarding based on strict routing rules. Maybe 
there is an option in that device at 192.168.160.44 to do loose routing, 
which is the recommended one in RFC3261 (strict routing is from the old 
rfc of SIP).

Regarding the missing '<' in the Route header sent out from Kamailio, 
there was an issue in the code handling this specific situation, should 
be fixed now by commit:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=cbb62f8619b513605498d00abc5d4c8b2f5654d7

You have to apply that patch or use the latest git branch 3.2. Let us 
know if works fine.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From vijay.thakur at loopmethods.com  Fri Aug  3 10:38:09 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Fri, 03 Aug 2012 14:08:09 +0530
Subject: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error
Message-ID: <501B8DF1.7030002@loopmethods.com>

Hello all,

I have configure Kamailio 3.1.5 Server. All things are working fine. 
When i make a call from Soft phone (X-Lite) to iphone, all is working 
fine. But in other case call from iphone to Softphone is not working, 
even not ringing. During checking the logs i am getting the error:

Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= 
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF 
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

This is server is hosted on Linnode.

Kindly guide me to solve the problem

-- 
Best Regards,

Vijay Thakur
(Assistant Manager - Networks)
Mobile   : +91 8744018065
Mail     : vijay.thakur at loopmethods.com

Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web: www.loopmethods.com

LOOP Disclaimer -------------------------------------------------------------------------------------------------
This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited.
-----------------------------------------------------------------------------------------------------------------



From miconda at gmail.com  Fri Aug  3 10:51:17 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Aug 2012 10:51:17 +0200
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <501B48C1.5020005@akan-tech.com>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
	<5019CADF.4070703@akan.net> <501A27D7.90107@gmail.com>
	<501B48C1.5020005@akan-tech.com>
Message-ID: <501B9105.3050508@gmail.com>

Hello,

On 8/3/12 5:42 AM, Nathaniel L Keeling wrote:
> Yes but it was for an older version that was uninstalled. I changed 
> the path to pull in the current version and still received the same 
> error. I checked the version for openssl and the same thing was 
> happening. I changed the path again and recompiled kamailio and still 
> no change. I noticed when compiling kamailio, the tls.so module was 
> complaining that the version of openssl was less then 1.0. The version 
> that I installed is 1.0.1c.
openssl libs can be installed with many versions at the same time, as 
the package names differ -- so 0.9x and 1.x can exist installed at the 
same time. Double check you don't have both installed, as some 
applications out there require 0.9, others 1.0, they could get installed 
due to dependencies.

It may happen that opensuse build service used a specific openssl 
version when building the rpms -- the rpm spec requires generic packages 
openssl and openssl-devel for building.

What is the version of your OS?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From mino.haluz at gmail.com  Fri Aug  3 15:53:34 2012
From: mino.haluz at gmail.com (Mino Haluz)
Date: Fri, 3 Aug 2012 15:53:34 +0200
Subject: [SR-Users] 302 redirect with 2 numbers registered
Message-ID: <CAFA7yDQRCtGMMeOgj7GgZhi=begA-AUcc56fgS73TFoGtYqPaQ@mail.gmail.com>

Hi,

one number is registered on 2 phones. Phone1 has Always redirect set
to another number. When incoming call is initiated, Phone2 is ringing
and Phone1 sends 302 to the proxy. However the proxy does not send 302
to the caller (for ex. GW), but it waits for timeout of the Phone2.
Then the proxy sends 302 to the caller.

Can I do in kamailio, that it will ring on the Phone1 and also on the
number where it is redirected? I know kamailio is a proxy and cannot
initiate a call, but is there any solution? Thanks.

Mino


From apogrebennyk at sipwise.com  Fri Aug  3 16:07:50 2012
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Fri, 03 Aug 2012 16:07:50 +0200
Subject: [SR-Users] 302 redirect with 2 numbers registered
In-Reply-To: <CAFA7yDQRCtGMMeOgj7GgZhi=begA-AUcc56fgS73TFoGtYqPaQ@mail.gmail.com>
References: <CAFA7yDQRCtGMMeOgj7GgZhi=begA-AUcc56fgS73TFoGtYqPaQ@mail.gmail.com>
Message-ID: <501BDB36.7050503@sipwise.com>

Mino, I am not sure, but you could try the following:
set failure_reply_mode 3
(http://kamailio.org/docs/modules/stable/modules/tm.html#failure_reply_mode),
then handle 302 redirect in the proxy and use the contact as a new
branch like this:

if(status == "302")
{
	$var(contact) = $ct;
	$var(contact) = $(var(contact){nameaddr.uri});
	$du = $var(contact);
	append_branch();
	t_relay();
}
Would it ring it while branch to Phone1 is still active? Maybe not, but
you will need to try..

On 08/03/2012 03:53 PM, Mino Haluz wrote:
> Hi,
> 
> one number is registered on 2 phones. Phone1 has Always redirect set
> to another number. When incoming call is initiated, Phone2 is ringing
> and Phone1 sends 302 to the proxy. However the proxy does not send 302
> to the caller (for ex. GW), but it waits for timeout of the Phone2.
> Then the proxy sends 302 to the caller.
> 
> Can I do in kamailio, that it will ring on the Phone1 and also on the
> number where it is redirected? I know kamailio is a proxy and cannot
> initiate a call, but is there any solution? Thanks.
> 
> Mino
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 



From evilzluk at gmail.com  Fri Aug  3 16:38:23 2012
From: evilzluk at gmail.com (Konstantin M.)
Date: Fri, 3 Aug 2012 17:38:23 +0300
Subject: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error
In-Reply-To: <501B8DF1.7030002@loopmethods.com>
References: <501B8DF1.7030002@loopmethods.com>
Message-ID: <CAEYtTzdeY88X-GC+wDwLou+smsB60h+M36h7xwCu448V0YtD2g@mail.gmail.com>

Looks like your firewall is passing a state RELATED,ESTABLISHED and don't
have a permit rule for a state NEW.
Check your iptables or simply do a command: iptables-save >
/tmp/iptables.txt    and paste this file (/tmp/iptables.txt) to pastebin

2012/8/3 Vijay Thakur <vijay.thakur at loopmethods.com>

> Hello all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine. When
> i make a call from Soft phone (X-Lite) to iphone, all is working fine. But
> in other case call from iphone to Softphone is not working, even not
> ringing. During checking the logs i am getting the error:
>
> Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:**75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP
> SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK
> URGP=0 OPT (0101080A000E20610932B25A)
>
> This is server is hosted on Linnode.
>
> Kindly guide me to solve the problem
>
> --
> Best Regards,
>
> Vijay Thakur
> (Assistant Manager - Networks)
> Mobile   : +91 8744018065
> Mail     : vijay.thakur at loopmethods.com
>
> Loop IT Methods Private Limited
> 1st Floor, B-10, Sector-7, Noida, (U.P) India
> Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178(AUS)
> Fax: +91 971 728 330
> Web: www.loopmethods.com
>
> LOOP Disclaimer ------------------------------**
> ------------------------------**------------------------------**-------
> This message (including any attachments) contains confidential information
> intended for a specific individual and purpose, and is protected by law. If
> you are not the intended recipient, you should delete this message and are
> hereby notified that any disclosure, copying, or distribution of this
> message, or the taking of any action based on it, is strictly prohibited.
> ------------------------------**------------------------------**
> ------------------------------**-----------------------
>
>
> ______________________________**_________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
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From henning at loca.net  Sat Aug  4 00:30:28 2012
From: henning at loca.net (Henning Holtschneider)
Date: Sat, 4 Aug 2012 00:30:28 +0200
Subject: [SR-Users] REGISTER replication to multiple hosts
Message-ID: <AE3A40A1-DC5C-4957-8BEA-A976FABB4621@loca.net>

Hello,

I'm trying to migrate an existing configuration from OpenSIPS 1.4.x to Kamailio 3.2.x. The old configuration replicates REGISTER requests to multiple servers using this code in the REGISTER routing block:

	add_sock_hdr("Local-Sock");
	add_rcv_param();
	append_branch("sip:1.1.1.1:5060");
	t_replicate("sip:2.2.2.2:5060");
	exit;

From what I read in the documentation, the behaviour of append_branch() is different in Kamailio, so I modified my code like this:

	add_sock_hdr("Local-Sock");
	add_rcv_param();
	append_branch();
	seturi(sip:1.1.1.1:5060");
	t_replicate("sip:2.2.2.2:5060");
	exit;

Unfortunately, I do not see any REGISTER requests arriving on 1.1.1.1 so I assume that I made a mistake when forking the request. Can anyone point me in the right direction?

Thanks,
Henning Holtschneider
--
LocaNet oHG - http://www.loca.net
Lindemannstrasse 81, D-44137 Dortmund
tel +49 231 91596-25, fax +49 231 91596-55
sip 25 at voip.loca.net

Registergericht Amtsgericht Dortmund HRA 14208
Gesch?ftsf?hrer Sven Haufe, Henning Holtschneider

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From edesibe at googlemail.com  Fri Aug  3 09:31:44 2012
From: edesibe at googlemail.com (Ivan Milivojevic)
Date: Fri, 3 Aug 2012 09:31:44 +0200
Subject: [SR-Users] Question about Proxy-Authorization header in ACK
Message-ID: <010001cd714a$092f3620$1b8da260$@com>

Hi all,
 
I have a one question regarding ACK and Proxy-Authorization header.
I am testing Kamailio 3.3 as SIP proxy,default config. I made one call with
2 phones where one has Public IP while other is behind NAT. SIP clients are
Panasonic KX-UT133 and other is 1-NET (ex Sweden Mobile&CDMA provider ,over
50M users) Everything is working just fine, both RTP and SIP. On the other
hand one of the clients,1-NET one, send Proxy-Authorization in ACK when he
receives 200 OK from Kamailio.
 
PUBLIC_IP_USER1 - user 1000
PUBLIC_IP_USER2 - user 1001
 
KAMAILIO ==> 1-NET    
    SIP/2.0 200 OKr\n
    Record-Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes>
KAMAILIO_PUBLIC_IP;lr=on;nat=yes> 
    Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport=5064;branch=z9hG4bK934894606

    Call-ID: 221374358 
    From: blabla <sip:1000@ <sip:1000 at KAMAILIO_PUBLIC_IP>
KAMAILIO_PUBLIC_IP>;tag=507511069 
    To: <sip:1001@ <sip:1001 at KAMAILIO_PUBLIC_IP:5060>
KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 
    CSeq: 21 INVITE 
    Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER 
    Contact: <sip:1001@ <sip:1001 at PUBLIC_IP_USER2:1027>
PUBLIC_IP_USER2:1027> 
    Require: replaces 
    Content-Type: application/sdp 
    Server: Panasonic_KX-UT133NE/01.081 (0080f0cedd83) 
    Content-Length: 182 
     
    v=0 
    o=- 1343736535 1343736535 IN IP4 KAMAILIO_PUBLIC_IP 
    s=- 
    c=IN IP4 KAMAILIO_PUBLIC_IP 
    t=0 0 
    m=audio 20412 RTP/AVP 18 
    a=rtpmap:18 G729/8000 
    a=sendrecv 
    a=ptime:20 
    a=nortpproxy:yes 
 
1-NET ==> KAMAILIO
    ACK sip:1001@ <sip:1001 at PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027
SIP/2.0 
    Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport;branch=z9hG4bK642028490 
    Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes>
KAMAILIO_PUBLIC_IP;lr=on;nat=yes> 
    From: blabla <sip:1000@ <sip:1000 at KAMAILIO_PUBLIC_IP>
KAMAILIO_PUBLIC_IP>;tag=507511069 
    To: <sip:1001@ <sip:1001 at KAMAILIO_PUBLIC_IP:5060>
KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 
    Call-ID: 221374358 
    CSeq: 21 ACK 
    Contact: <sip:1000@ <sip:1000 at PUBLIC_IP_USER1:5064>
PUBLIC_IP_USER1:5064> 
    [truncated] Proxy-Authorization: Digest username="1000",
realm="KAMAILIO_PUBLIC_IP", nonce="UBfL8lAXysb5tJCs80ZnthyPl9IzmRZk",
uri="sip:1001 at KAMAILIO_PUBLIC_IP:5060",
response="7824519cdad9f1c2c79027a2d7522344", algorithm=MD5,
cnonce="0a4f113b", q
    Max-Forwards: 70 
    User-Agent: Serbia_2.00 
    Content-Length: 0 
     
I attach txt file with call flow, I can send pcap also.
 
I think that the issue is related with bad client but I need another
opinion.
Does anyone has an idea about this issue?
 
Best Regards,
Ivan
 
 
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From techsup0073 at gmail.com  Sat Aug  4 13:28:46 2012
From: techsup0073 at gmail.com (Techie Sup)
Date: Sat, 4 Aug 2012 16:58:46 +0530
Subject: [SR-Users] RTP over TCP
Message-ID: <CABA-duT2NRMek9PqWaH7UE5Pe659vKEqN083rRX4CoJvMCvaew@mail.gmail.com>

Hi All,

Is there any possibility of sending RTP over TCP, because we are building a
softphone based on PJSIP, but seems that some ISP are blocking RTP packets
over UDP. Any solutions for this.

Regards
Gloria
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From techsup0073 at gmail.com  Sun Aug  5 11:06:07 2012
From: techsup0073 at gmail.com (Techie Sup)
Date: Sun, 5 Aug 2012 14:36:07 +0530
Subject: [SR-Users] RTP over TCP
In-Reply-To: <CABA-duT2NRMek9PqWaH7UE5Pe659vKEqN083rRX4CoJvMCvaew@mail.gmail.com>
References: <CABA-duT2NRMek9PqWaH7UE5Pe659vKEqN083rRX4CoJvMCvaew@mail.gmail.com>
Message-ID: <CABA-duQbWfkSTRBybYVT25Ku9SZWK_NSOhAcLMmRAS2Z944FeQ@mail.gmail.com>

---------- Forwarded message ----------
From: Techie Sup <techsup0073 at gmail.com>
Date: Sat, 4 Aug 2012 16:58:46 +0530
Subject: RTP over TCP
To: sr-users at lists.sip-router.org

Hi All,

Is there any possibility of sending RTP over TCP, because we are building a
softphone based on PJSIP, but seems that some ISP are blocking RTP packets
over UDP. Any solutions for this.

Regards
Gloria


From abalashov at evaristesys.com  Sun Aug  5 11:13:48 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sun, 05 Aug 2012 05:13:48 -0400
Subject: [SR-Users] RTP over TCP
In-Reply-To: <CABA-duQbWfkSTRBybYVT25Ku9SZWK_NSOhAcLMmRAS2Z944FeQ@mail.gmail.com>
References: <CABA-duT2NRMek9PqWaH7UE5Pe659vKEqN083rRX4CoJvMCvaew@mail.gmail.com>
	<CABA-duQbWfkSTRBybYVT25Ku9SZWK_NSOhAcLMmRAS2Z944FeQ@mail.gmail.com>
Message-ID: <501E394C.6090306@evaristesys.com>

Yes, if you just keep forwarding your message to the list every ~22 
hours, that will surely elicit a response.

On 08/05/2012 05:06 AM, Techie Sup wrote:

> ---------- Forwarded message ----------
> From: Techie Sup <techsup0073 at gmail.com>
> Date: Sat, 4 Aug 2012 16:58:46 +0530
> Subject: RTP over TCP
> To: sr-users at lists.sip-router.org
>
> Hi All,
>
> Is there any possibility of sending RTP over TCP, because we are building a
> softphone based on PJSIP, but seems that some ISP are blocking RTP packets
> over UDP. Any solutions for this.
>
> Regards
> Gloria
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From abalashov at evaristesys.com  Sun Aug  5 11:19:44 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sun, 05 Aug 2012 05:19:44 -0400
Subject: [SR-Users] RTP over TCP
In-Reply-To: <CABA-duT2NRMek9PqWaH7UE5Pe659vKEqN083rRX4CoJvMCvaew@mail.gmail.com>
References: <CABA-duT2NRMek9PqWaH7UE5Pe659vKEqN083rRX4CoJvMCvaew@mail.gmail.com>
Message-ID: <501E3AB0.5030402@evaristesys.com>

There is that possibility, yes, insofar as the RFC 3550 does contemplate 
non-UDP transports.

The real question is: why are you asking this list?  Kamailio/sip-router 
is not involved in RTP transport.  Why would you look for solutions here?

On 08/04/2012 07:28 AM, Techie Sup wrote:

> Hi All,
>
> Is there any possibility of sending RTP over TCP, because we are
> building a softphone based on PJSIP, but seems that some ISP are
> blocking RTP packets over UDP. Any solutions for this.
>
> Regards
> Gloria
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From universe at truemetal.org  Sun Aug  5 12:47:23 2012
From: universe at truemetal.org (Markus)
Date: Sun, 05 Aug 2012 12:47:23 +0200
Subject: [SR-Users] Getting started with Kamailio
In-Reply-To: <501AFAFA.70405@evaristesys.com>
References: <501AAD9A.3030406@truemetal.org> <501AFAFA.70405@evaristesys.com>
Message-ID: <501E4F3B.30602@truemetal.org>

Hi Alex,

Am 03.08.2012 00:11, schrieb Alex Balashov:
>> - Give only a single IP address to all customers and termination
>> providers (the same IP address), 195.5.5.5.
>
> That would be difficult, unless you really have your anycast ducks in a
> row.  A more common way to approach this problem is SRV records, though
> not all SIP endpoints out there implement SRV correctly or make use of
> it in the manner contemplated by a redundancy-oriented use of them.

I think I'll be fine if I prepend my ASN several times at one location 
(Spain or France). I will probably still get some tiny traffic from the 
direct upstream and maybe its customers, if they have configured it this 
way, but the traffic that is originating from my IPs at this location 
will also get routed via the direct upstream, so everything should be in 
order. Therefore that location will act as backup only (except the 
constant tiny traffic that it will probably receive).

I'll cut the rest of your answers as I have nothing to add. Just wanted 
to say a big thank you for the detailed answers! That helped.

Regards
Markus




From noreply+2542793023 at badoo.com  Sun Aug  5 17:37:07 2012
From: noreply+2542793023 at badoo.com (Badoo)
Date: Sun,  5 Aug 2012 15:37:07 +0000
Subject: [SR-Users] Gorka left a message for you
Message-ID: <20120805153708.7B6F1370540@mail.iptel.org>

Gorka left a message for you

Only you can see the sender and content of your message, and you can delete it anytime. You can instantly reply using our message exchange system:
http://eu1.badoo.com/0106730996/in/UJF4Z.6PYhQ/?lang_id=3&m=63&mid=501e9322000000000003000006c28ba2000100970652

Some other people in the area who are on Badoo
Ali (Jeddah, Saudi Arabia)
Fnoo meni jnoo (Jeddah, Saudi Arabia)
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http://eu1.badoo.com/0106730996/in/UJF4Z.6PYhQ/?lang_id=3&m=63&mid=501e9322000000000003000006c28ba2000100970652

If the link in this message does not work, try copying and pasting it into your browser.

This email is part of our delivery procedure for the message sent by Gorka. If you have received this email by mistake, please ignore it. The message will be deleted soon.

Have fun!
The Badoo Team

You have received this email from Badoo Trading Limited (postal address below).
http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=serusers%40iptel.org&block_code=8f4b94&m=63&mid=501e9322000000000003000006c28ba2000100970652
Badoo Trading Limited is a limited company registered in England and Wales
under CRN 7540255 with its registered office at 12 Red Lion Square, London, WC1R 4QD.
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From vijay.thakur at loopmethods.com  Mon Aug  6 06:53:23 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Mon, 06 Aug 2012 10:23:23 +0530
Subject: [SR-Users] sr-users Digest, Vol 87, Issue 9
In-Reply-To: <mailman.4028.1344061497.2421.sr-users@lists.sip-router.org>
References: <mailman.4028.1344061497.2421.sr-users@lists.sip-router.org>
Message-ID: <501F4DC3.8020106@loopmethods.com>

Hi Konstantin,


Thanks for your kind reply. You can check the iptables output at : http://pastebin.com/i3zUfVeb.
I hope that this will give you enough clue in right direction.

With Regards,

Vijay Thakur


======================================================================================================================

Message: 3
Date: Fri, 3 Aug 2012 17:38:23 +0300
From: "Konstantin M."<evilzluk at gmail.com>
Subject: Re: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0
	Error
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
	Users	Mailing List"<sr-users at lists.sip-router.org>
Message-ID:
	<CAEYtTzdeY88X-GC+wDwLou+smsB60h+M36h7xwCu448V0YtD2g at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Looks like your firewall is passing a state RELATED,ESTABLISHED and don't
have a permit rule for a state NEW.
Check your iptables or simply do a command: iptables-save >
/tmp/iptables.txt    and paste this file (/tmp/iptables.txt) to pastebin

2012/8/3 Vijay Thakur<vijay.thakur at loopmethods.com>

> Hello all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine. When
> i make a call from Soft phone (X-Lite) to iphone, all is working fine. But
> in other case call from iphone to Softphone is not working, even not
> ringing. During checking the logs i am getting the error:
>
> Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:**75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP
> SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK
> URGP=0 OPT (0101080A000E20610932B25A)
>
> This is server is hosted on Linnode.
>
> Kindly guide me to solve the problem
>
> --
> Best Regards,
>
> Vijay Thakur
> (Assistant Manager - Networks)
> Mobile   : +91 8744018065
> Mail     :vijay.thakur at loopmethods.com


From miconda at gmail.com  Mon Aug  6 08:45:31 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Aug 2012 08:45:31 +0200
Subject: [SR-Users] REGISTER replication to multiple hosts
In-Reply-To: <AE3A40A1-DC5C-4957-8BEA-A976FABB4621@loca.net>
References: <AE3A40A1-DC5C-4957-8BEA-A976FABB4621@loca.net>
Message-ID: <501F680B.8050009@gmail.com>

Hello,

seturi() just sets the request uri, then the t_replicate function you 
call is sending to 2.2.2.2.

You can use km_append_branch(...) which is the function to append 
branches inherited from openser times. It is in the kex module:

http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404

Just fyi, there is an alternative of distributing the location records 
using the presence mechanism, based on an IETF RFC, a recommendation for 
IMS networks -- see pua_reginfo and presence_reginfo

http://kamailio.org/docs/modules/stable/modules_k/pua_reginfo.html
http://kamailio.org/docs/modules/stable/modules_k/presence_reginfo.html

Still the t_replicate() should be the simpler approach and should work 
just fine with latest version.

Cheers,
Daniel

On 8/4/12 12:30 AM, Henning Holtschneider wrote:
> Hello,
>
> I'm trying to migrate an existing configuration from OpenSIPS 1.4.x to Kamailio 3.2.x. The old configuration replicates REGISTER requests to multiple servers using this code in the REGISTER routing block:
>
> 	add_sock_hdr("Local-Sock");
> 	add_rcv_param();
> 	append_branch("sip:1.1.1.1:5060");
> 	t_replicate("sip:2.2.2.2:5060");
> 	exit;
>
>  From what I read in the documentation, the behaviour of append_branch() is different in Kamailio, so I modified my code like this:
>
> 	add_sock_hdr("Local-Sock");
> 	add_rcv_param();
> 	append_branch();
> 	seturi(sip:1.1.1.1:5060");
> 	t_replicate("sip:2.2.2.2:5060");
> 	exit;
>
> Unfortunately, I do not see any REGISTER requests arriving on 1.1.1.1 so I assume that I made a mistake when forking the request. Can anyone point me in the right direction?
>
> Thanks,
> Henning Holtschneider
> --
> LocaNet oHG - http://www.loca.net
> Lindemannstrasse 81, D-44137 Dortmund
> tel +49 231 91596-25, fax +49 231 91596-55
> sip 25 at voip.loca.net
>
> Registergericht Amtsgericht Dortmund HRA 14208
> Gesch?ftsf?hrer Sven Haufe, Henning Holtschneider
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From klaus.mailinglists at pernau.at  Mon Aug  6 09:50:02 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Mon, 06 Aug 2012 09:50:02 +0200
Subject: [SR-Users] Question about Proxy-Authorization header in ACK
In-Reply-To: <010001cd714a$092f3620$1b8da260$@com>
References: <010001cd714a$092f3620$1b8da260$@com>
Message-ID: <501F772A.5090901@pernau.at>

Authenticating ACK is troublesome as ACK may not be challenged. Thus, 
some clients (IIRC also eyebam does it) send ACK with 
Proxy-Authentication using previous nonce to avoid troubles in case the 
proxy authenticates the ACK too.

regards
Klaus

On 03.08.2012 09:31, Ivan Milivojevic wrote:
> Hi all,
>
> I have a one question regarding ACK and Proxy-Authorization header.
>
> I am testing Kamailio 3.3 as SIP proxy,default config. I made one call
> with 2 phones where one has Public IP while other is behind NAT. SIP
> clients are Panasonic KX-UT133 and other is 1-NET (ex Sweden Mobile&CDMA
> provider ,over 50M users) Everything is working just fine, both RTP and
> SIP. On the other hand one of the clients,1-NET one, send
> Proxy-Authorization in ACK when he receives 200 OK from Kamailio.
>
> *PUBLIC_IP_USER1 ? user 1000*
>
> *PUBLIC_IP_USER2 ? user 1001*
>
> KAMAILIO ? 1-NET
>
> SIP/2.0 200 OKr\n
>
> Record-Route: <sip:*KAMAILIO_PUBLIC_IP*;lr=on;nat=yes
> <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes>>
>
> Via: SIP/2.0/UDP *PUBLIC_IP_USER1*:5064;rport=5064;branch=z9hG4bK934894606
>
> Call-ID: 221374358
>
> From: blabla <sip:1000@*KAMAILIO_PUBLIC_IP*
> <sip:1000 at KAMAILIO_PUBLIC_IP>>;tag=507511069
>
> To: <sip:1001@*KAMAILIO_PUBLIC_IP*:5060
> <sip:1001 at KAMAILIO_PUBLIC_IP:5060>>;tag=2148378512
>
> CSeq: 21 INVITE
>
> Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER
>
> Contact: <sip:1001@*PUBLIC_IP_USER2*:1027 <sip:1001 at PUBLIC_IP_USER2:1027>>
>
> Require: replaces
>
> Content-Type: application/sdp
>
> Server: Panasonic_KX-UT133NE/01.081 (0080f0cedd83)
>
> Content-Length: 182
>
> v=0
>
> o=- 1343736535 1343736535 IN IP4 KAMAILIO_PUBLIC_IP
>
> s=-
>
> c=IN IP4 KAMAILIO_PUBLIC_IP
>
> t=0 0
>
> m=audio 20412 RTP/AVP 18
>
> a=rtpmap:18 G729/8000
>
> a=sendrecv
>
> a=ptime:20
>
> a=nortpproxy:yes
>
> 1-NET ? KAMAILIO
>
> ACK sip:1001@*PUBLIC_IP_USER2*:1027 <sip:1001 at PUBLIC_IP_USER2:1027> SIP/2.0
>
> Via: SIP/2.0/UDP *PUBLIC_IP_USER1*:5064;rport;branch=z9hG4bK642028490
>
> Route: <sip:*KAMAILIO_PUBLIC_IP*;lr=on;nat=yes
> <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes>>
>
> From: blabla <sip:1000@*KAMAILIO_PUBLIC_IP*
> <sip:1000 at KAMAILIO_PUBLIC_IP>>;tag=507511069
>
> To: <sip:1001@*KAMAILIO_PUBLIC_IP*:5060
> <sip:1001 at KAMAILIO_PUBLIC_IP:5060>>;tag=2148378512
>
> Call-ID: 221374358
>
> CSeq: 21 ACK
>
> Contact: <sip:1000@*PUBLIC_IP_USER1*:5064 <sip:1000 at PUBLIC_IP_USER1:5064>>
>
> *[truncated] Proxy-Authorization: Digest username="1000",
> realm="KAMAILIO_PUBLIC_IP", nonce="UBfL8lAXysb5tJCs80ZnthyPl9IzmRZk",
> uri="sip:1001 at KAMAILIO_PUBLIC_IP:5060",
> response="7824519cdad9f1c2c79027a2d7522344", algorithm=MD5,
> cnonce="0a4f113b", q*
>
> Max-Forwards: 70
>
> User-Agent: Serbia_2.00
>
> Content-Length: 0
>
> I attach txt file with call flow, I can send pcap also.
>
> I think that the issue is related with bad client but I need another
> opinion.
>
> Does anyone has an idea about this issue?
>
> Best Regards,
>
> Ivan
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From phillman25 at gmail.com  Mon Aug  6 10:24:03 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 6 Aug 2012 11:24:03 +0300
Subject: [SR-Users] B2BUA issues
Message-ID: <CAEprJzJu8FP5NmgM7keCtHYk4A-dkMPOw41e7kWhJmpfHHYBDA@mail.gmail.com>

Dear List

I am trying to accomplish the following:


Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx)  ==> Cisco PGW
2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> Kamailio(xxx.xxx.xxx.xxx) ==>
Asterisk PABX (192.168.10.189)

When trying the above scenario, the call is silent and drops after a few
seconds. In syslog i observe the following error:

*ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
name-addr (sip:22030305 at 192.168.10.189:5060>)*

Looking at the sip trace i see that his might be caused by the ACK message
received from the ASTERISK PABX? :

ACK sip:22030305 at 192.168.10.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
Route:
<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
yyy.yyy.yyy.yyy
;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
Max-Forwards: 70
From: "22498045" <sip:22498045 at 192.168.10.189>;tag=as166b1eea
To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
Contact: <sip:22498045 at 192.168.10.189:5060>
Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.12.0)
Content-Length: 0

After contacting Cisco they informed us that issue is cause by B2BUA from
our current version of Cisco PGW 2200 that doesn't support this feature. Is
there a module, solution that i can implement on Kamailio that could
temporarily resolve this issue?

Thanking you in advance.

Phillip
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From abalashov at evaristesys.com  Mon Aug  6 10:26:28 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 06 Aug 2012 04:26:28 -0400
Subject: [SR-Users] B2BUA issues
In-Reply-To: <CAEprJzJu8FP5NmgM7keCtHYk4A-dkMPOw41e7kWhJmpfHHYBDA@mail.gmail.com>
References: <CAEprJzJu8FP5NmgM7keCtHYk4A-dkMPOw41e7kWhJmpfHHYBDA@mail.gmail.com>
Message-ID: <501F7FB4.8040700@evaristesys.com>

What is the larger objective?  Are you using the PGW purely as a B2BUA? 
  If so, that's a colossally overblown waste of resources;  just use 
something like SEMS or Asterisk.

On 08/06/2012 04:24 AM, phillman25 wrote:

> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx)  ==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305 at 192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305 at 192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045 at 192.168.10.189
> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045 at 192.168.10.189:5060
> <http://sip:22498045 at 192.168.10.189:5060>>
> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From vijay.thakur at loopmethods.com  Mon Aug  6 12:08:30 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Mon, 06 Aug 2012 15:38:30 +0530
Subject: [SR-Users] Kernel Droping SIP packet
Message-ID: <501F979E.30607@loopmethods.com>

Hi all,

I have configure Kamailio 3.1.5 Server. All things are working fine. 
When i make a call from Soft phone (X-Lite) to iphone, all is working 
fine. But in other case call from iphone to Softphone is not working, 
even not ringing. During checking the logs i am getting the error:

Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= 
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF 
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

I have not implemented any firewall. You can check the out put of my 
iptables : http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.

With thanks in advance. Sorry dual posting.

Vijay TH


From phillman25 at gmail.com  Mon Aug  6 12:17:23 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 6 Aug 2012 13:17:23 +0300
Subject: [SR-Users] B2BUA issues
Message-ID: <CAEprJzJNzXEkODp_wMxBU5fedN0mPrQbBSWgQnBBsig0L-K9Jg@mail.gmail.com>

Hi Alex

Thanks for your prompt reply.

The PGW 2200 solution is used as our core PSTN gateway where its currently
handling many SS7, H.323 and SIP interconnections. However, there are a few
scenarios like the example described below, that the call is originating
from Kamailio being sent to the PGW and then back to Kamailio for
termination and this scenario doesn't seem to work.

Do you think that by implementing SEMS or Asterisk in between the PGW and
Kamailio could resolve this issue for these specific scenarios?

From abalashov at evaristesys.com  Mon Aug  6 12:33:07 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 06 Aug 2012 06:33:07 -0400
Subject: [SR-Users] B2BUA issues
Message-ID: <mae3fo9qpnr9dd67cf7b560f.1344249187478@email.android.com>

The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported.?

Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA.




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/phillman25 <phillman25 at gmail.com> wrote:Hi Alex

Thanks for your prompt reply.

The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work.

Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios?
From your experience what do you think would be a better solution?

Thanks again!
Phillip

========================
Message: 2
Date: Mon, 06 Aug 2012 04:26:28 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [SR-Users] B2BUA issues
To:?sr-users at lists.sip-router.org
Message-ID: <501F7FB4.8040700 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

What is the larger objective? ?Are you using the PGW purely as a B2BUA?
? If so, that's a colossally overblown waste of resources; ?just use
something like SEMS or Asterisk.

On 08/06/2012 04:24 AM, phillman25 wrote:

> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ?==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305 at 192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK?sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305 at 192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045 at 192.168.10.189
> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045 at 192.168.10.189:5060
> <http://sip:22498045 at 192.168.10.189:5060>>
> Call-ID:?5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip
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From abalashov at evaristesys.com  Mon Aug  6 12:35:04 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 06 Aug 2012 06:35:04 -0400
Subject: [SR-Users] Kernel Droping SIP packet
Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618@email.android.com>

You might consider pasting the actual output of: iptables -L -n

This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> wrote:Hi all,

I have configure Kamailio 3.1.5 Server. All things are working fine. 
When i make a call from Soft phone (X-Lite) to iphone, all is working 
fine. But in other case call from iphone to Softphone is not working, 
even not ringing. During checking the logs i am getting the error:

Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= 
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF 
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

I have not implemented any firewall. You can check the out put of my 
iptables : http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.

With thanks in advance. Sorry dual posting.

Vijay TH

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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From phillman25 at gmail.com  Mon Aug  6 13:57:51 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 6 Aug 2012 14:57:51 +0300
Subject: [SR-Users] B2BUA issues
In-Reply-To: <mae3fo9qpnr9dd67cf7b560f.1344249187478@email.android.com>
References: <mae3fo9qpnr9dd67cf7b560f.1344249187478@email.android.com>
Message-ID: <CAEprJz+20Wj3YaLZ1=dF9cYxB1yMJphWmf3s+RcvGj2WTpPGOw@mail.gmail.com>

Hello Alex

Will try with SEMS first, found something called sip:provider CE v2.4 from
http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm not
mistaken, this seems to combine Kamailio with SEMS? Do you think that this
might be an easier installation rather than installing SEMS on its own as
it seems to provide more documentation?

Thanks again!



On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> The short answer to your latter question is: yes. Cisco media and PSTN
> gateways have never hairpinned SIP-to-SIP calls well, even when officially
> supported.
>
> Asterisk has a lower learning curve due to the abundance of information
> and tutorials, but SEMS would make more sense, since all you need is a
> signaling B2BUA.
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> phillman25 <phillman25 at gmail.com> wrote:
> Hi Alex
>
> Thanks for your prompt reply.
>
> The PGW 2200 solution is used as our core PSTN gateway where its currently
> handling many SS7, H.323 and SIP interconnections. However, there are a few
> scenarios like the example described below, that the call is originating
> from Kamailio being sent to the PGW and then back to Kamailio for
> termination and this scenario doesn't seem to work.
>
> Do you think that by implementing SEMS or Asterisk in between the PGW and
> Kamailio could resolve this issue for these specific scenarios?
> From your experience what do you think would be a better solution?
>
> Thanks again!
> Phillip
>
> ========================
> Message: 2
> Date: Mon, 06 Aug 2012 04:26:28 -0400
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [SR-Users] B2BUA issues
> To: sr-users at lists.sip-router.org
> Message-ID: <501F7FB4.8040700 at evaristesys.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> What is the larger objective?  Are you using the PGW purely as a B2BUA?
>   If so, that's a colossally overblown waste of resources;  just use
> something like SEMS or Asterisk.
>
> On 08/06/2012 04:24 AM, phillman25 wrote:
>
> > Dear List
> >
> > I am trying to accomplish the following:
> >
> >
> > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx)  ==> Cisco
> > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
> >
> > When trying the above scenario, the call is silent and drops after a few
> > seconds. In syslog i observe the following error:
> >
> > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> > name-addr (sip:22030305 at 192.168.10.189:5060
> > <http://sip:22030305 at 192.168.10.189:5060>>)*
> >
> > Looking at the sip trace i see that his might be caused by the ACK
> > message received from the ASTERISK PABX? :
> >
> > ACK sip:22030305 at 192.168.10.189:5060
> > <http://sip:22030305 at 192.168.10.189:5060> SIP/2.0
> > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> > Route:
> > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.
> 66623da5>,<sip:22030305@
> > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;
> lr=on;ftag=as166b1eea>
> > Max-Forwards: 70
> > From: "22498045" <sip:22498045 at 192.168.10.189
> > <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as166b1eea
> > To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
> > Contact: <sip:22498045 at 192.168.10.189:5060
> > <http://sip:22498045 at 192.168.10.189:5060>>
> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
> > <http://5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060>
> > CSeq: 102 ACK
> > User-Agent: FPBX-2.8.1(1.8.12.0)
> > Content-Length: 0
> >
> >
> > After contacting Cisco they informed us that issue is cause by B2BUA
> > from our current version of Cisco PGW 2200 that doesn't support this
> > feature. Is there a module, solution that i can implement on Kamailio
> > that could temporarily resolve this issue?
> >
> > Thanking you in advance.
> >
> > Phillip
>
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From abalashov at evaristesys.com  Mon Aug  6 14:05:12 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 06 Aug 2012 08:05:12 -0400
Subject: [SR-Users] B2BUA issues
Message-ID: <5a1cblsgm692cf4h7q8b8mmv.1344254712317@email.android.com>

For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se.?




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/phillman25 <phillman25 at gmail.com> wrote:Hello Alex

Will try with SEMS first, found something called?sip:provider CE v2.4 from?http://www.sipwise.com/news/announcements/spce-v2_4-release/?if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation?

Thanks again!



On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported.?

Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA.




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/

phillman25 <phillman25 at gmail.com> wrote:
Hi Alex

Thanks for your prompt reply.

The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work.

Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios?
From your experience what do you think would be a better solution?

Thanks again!
Phillip

========================
Message: 2
Date: Mon, 06 Aug 2012 04:26:28 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [SR-Users] B2BUA issues
To:?sr-users at lists.sip-router.org
Message-ID: <501F7FB4.8040700 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

What is the larger objective? ?Are you using the PGW purely as a B2BUA?
? If so, that's a colossally overblown waste of resources; ?just use
something like SEMS or Asterisk.

On 08/06/2012 04:24 AM, phillman25 wrote:

> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ?==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305 at 192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK?sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305 at 192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045 at 192.168.10.189
> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045 at 192.168.10.189:5060
> <http://sip:22498045 at 192.168.10.189:5060>>
> Call-ID:?5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip

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From vijay.thakur at loopmethods.com  Mon Aug  6 14:37:28 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Mon, 06 Aug 2012 18:07:28 +0530
Subject: [SR-Users] Kernel Droping SIP packet
Message-ID: <501FBA88.9030303@loopmethods.com>

Thanks for prompt reply.  Here is the out put if command iptables -L -n:

*Chain INPUT (policy ACCEPT)
target     prot opt source               destination

Chain FORWARD (policy ACCEPT)
target     prot opt source               destination

Chain OUTPUT (policy ACCEPT)
target     prot opt source               destination

Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.


Vijay TH

-==============================================================================

*

Date: Mon, 06 Aug 2012 06:35:04 -0400
From: Alex Balashov<abalashov at evaristesys.com>
Subject: Re: [SR-Users] Kernel Droping SIP packet
To:sr-users at lists.sip-router.org
Message-ID:<17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com>
Content-Type: text/plain; charset="utf-8"

You might consider pasting the actual output of: iptables -L -n

This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web:http://www.evaristesys.com/Vijay  Thakur<vijay.thakur at loopmethods.com>  wrote:Hi all,

I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking the logs i am getting the error:

Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

I have not implemented any firewall. You can check the out put of my
iptables :http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.

With thanks in advance. Sorry dual posting.

Vijay TH


*
*
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From phillman25 at gmail.com  Mon Aug  6 14:59:57 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 6 Aug 2012 15:59:57 +0300
Subject: [SR-Users] B2BUA issues
In-Reply-To: <5a1cblsgm692cf4h7q8b8mmv.1344254712317@email.android.com>
References: <5a1cblsgm692cf4h7q8b8mmv.1344254712317@email.android.com>
Message-ID: <CAEprJzJtfjXzCWV2Wu_8-jVretayyARXVZi1BKgXeF_gR_Q7-g@mail.gmail.com>

Ok Alex thanks for the info!

On Mon, Aug 6, 2012 at 3:05 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> For your relatively narrow, specific applications in your topology, no.
> You'd be better off installing SEMS per se.
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> phillman25 <phillman25 at gmail.com> wrote:
> Hello Alex
>
> Will try with SEMS first, found something called sip:provider CE v2.4
> from http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm
> not mistaken, this seems to combine Kamailio with SEMS? Do you think that
> this might be an easier installation rather than installing SEMS on its own
> as it seems to provide more documentation?
>
> Thanks again!
>
>
>
> On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov at evaristesys.com>wrote:
>
>> The short answer to your latter question is: yes. Cisco media and PSTN
>> gateways have never hairpinned SIP-to-SIP calls well, even when officially
>> supported.
>>
>> Asterisk has a lower learning curve due to the abundance of information
>> and tutorials, but SEMS would make more sense, since all you need is a
>> signaling B2BUA.
>>
>>
>>
>>
>> -- Alex
>>
>> --
>> Sent from my Samsung mobile, and thus lacking in the refinement one might
>> expect from a proper keyboard.
>>
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 235 E Ponce de Leon Ave
>> Suite 106
>> Decatur, GA 30030
>> Tel: +1-678-954-0670
>> Web: http://www.evaristesys.com/
>>
>> phillman25 <phillman25 at gmail.com> wrote:
>> Hi Alex
>>
>> Thanks for your prompt reply.
>>
>> The PGW 2200 solution is used as our core PSTN gateway where its
>> currently handling many SS7, H.323 and SIP interconnections. However, there
>> are a few scenarios like the example described below, that the call is
>> originating from Kamailio being sent to the PGW and then back to Kamailio
>> for termination and this scenario doesn't seem to work.
>>
>> Do you think that by implementing SEMS or Asterisk in between the PGW and
>> Kamailio could resolve this issue for these specific scenarios?
>> From your experience what do you think would be a better solution?
>>
>> Thanks again!
>> Phillip
>>
>> ========================
>> Message: 2
>> Date: Mon, 06 Aug 2012 04:26:28 -0400
>> From: Alex Balashov <abalashov at evaristesys.com>
>> Subject: Re: [SR-Users] B2BUA issues
>> To: sr-users at lists.sip-router.org
>> Message-ID: <501F7FB4.8040700 at evaristesys.com>
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>> What is the larger objective?  Are you using the PGW purely as a B2BUA?
>>   If so, that's a colossally overblown waste of resources;  just use
>> something like SEMS or Asterisk.
>>
>> On 08/06/2012 04:24 AM, phillman25 wrote:
>>
>> > Dear List
>> >
>> > I am trying to accomplish the following:
>> >
>> >
>> > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx)  ==> Cisco
>> > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
>> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>> >
>> > When trying the above scenario, the call is silent and drops after a few
>> > seconds. In syslog i observe the following error:
>> >
>> > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
>> > name-addr (sip:22030305 at 192.168.10.189:5060
>> > <http://sip:22030305 at 192.168.10.189:5060>>)*
>> >
>> > Looking at the sip trace i see that his might be caused by the ACK
>> > message received from the ASTERISK PABX? :
>> >
>> > ACK sip:22030305 at 192.168.10.189:5060
>> > <http://sip:22030305 at 192.168.10.189:5060> SIP/2.0
>> > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
>> > Route:
>> > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.
>> 66623da5>,<sip:22030305@
>> > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;
>> lr=on;ftag=as166b1eea>
>> > Max-Forwards: 70
>> > From: "22498045" <sip:22498045 at 192.168.10.189
>> > <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as166b1eea
>> > To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
>> > Contact: <sip:22498045 at 192.168.10.189:5060
>> > <http://sip:22498045 at 192.168.10.189:5060>>
>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
>> > <http://5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060>
>> > CSeq: 102 ACK
>> > User-Agent: FPBX-2.8.1(1.8.12.0)
>> > Content-Length: 0
>> >
>> >
>> > After contacting Cisco they informed us that issue is cause by B2BUA
>> > from our current version of Cisco PGW 2200 that doesn't support this
>> > feature. Is there a module, solution that i can implement on Kamailio
>> > that could temporarily resolve this issue?
>> >
>> > Thanking you in advance.
>> >
>> > Phillip
>>
>
>
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From rfuchs at sipwise.com  Mon Aug  6 15:01:03 2012
From: rfuchs at sipwise.com (Richard Fuchs)
Date: Mon, 06 Aug 2012 09:01:03 -0400
Subject: [SR-Users] Kernel Droping SIP packet
In-Reply-To: <501F979E.30607@loopmethods.com>
References: <501F979E.30607@loopmethods.com>
Message-ID: <501FC00F.6000604@sipwise.com>

On 08/06/12 06:08, Vijay Thakur wrote:
> Hi all,
> 
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
> 
> Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

This is coming from nf_conntrack_sip, which is a netfilter connection
tracking kernel module for SIP. I've never used it, but judging from
what Google brings up, it seems to be very buggy. You should be able to
just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
and/or if you want to keep it from auto-loading, you can blacklist it in
/etc/modprobe.d/ and then reboot.

HTH

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From abalashov at evaristesys.com  Mon Aug  6 14:42:57 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 06 Aug 2012 08:42:57 -0400
Subject: [SR-Users] Kernel Droping SIP packet
Message-ID: <turqvr4fgiqrutc5pk4nan43.1344256977848@email.android.com>

Can you paste this output too??

lsmod | grep -i sip




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> wrote:Thanks for prompt reply.? Here is the out put if command iptables -L -n:

Chain INPUT (policy ACCEPT)
target???? prot opt source?????????????? destination???????? 

Chain FORWARD (policy ACCEPT)
target???? prot opt source?????????????? destination???????? 

Chain OUTPUT (policy ACCEPT)
target???? prot opt source?????????????? destination?? 

Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.


Vijay TH

-==============================================================================

 Date: Mon, 06 Aug 2012 06:35:04 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [SR-Users] Kernel Droping SIP packet
To: sr-users at lists.sip-router.org
Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com>
Content-Type: text/plain; charset="utf-8"

You might consider pasting the actual output of: iptables -L -n

This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> wrote:Hi all,

I have configure Kamailio 3.1.5 Server. All things are working fine. 
When i make a call from Soft phone (X-Lite) to iphone, all is working 
fine. But in other case call from iphone to Softphone is not working, 
even not ringing. During checking the logs i am getting the error:

Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= 
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF 
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

I have not implemented any firewall. You can check the out put of my 
iptables : http://pastebin.com/i3zUfVeb
The SIP server is hosted at linnode.

With thanks in advance. Sorry dual posting.

Vijay TH


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From evilzluk at gmail.com  Mon Aug  6 15:53:36 2012
From: evilzluk at gmail.com (Konstantin M.)
Date: Mon, 6 Aug 2012 16:53:36 +0300
Subject: [SR-Users] Kernel Droping SIP packet
In-Reply-To: <turqvr4fgiqrutc5pk4nan43.1344256977848@email.android.com>
References: <turqvr4fgiqrutc5pk4nan43.1344256977848@email.android.com>
Message-ID: <CAEYtTzeRjRK0Lks+7TWnTMP_5R5vn708Sv+um+U3VLbgj59ROw@mail.gmail.com>

rmmod nf_conntrack_sip

2012/8/6 Alex Balashov <abalashov at evaristesys.com>

> Can you paste this output too?
>
> lsmod | grep -i sip
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> Vijay Thakur <vijay.thakur at loopmethods.com> wrote:
> Thanks for prompt reply.  Here is the out put if command iptables -L -n:
>
> *Chain INPUT (policy ACCEPT)
> target     prot opt source               destination
>
> Chain FORWARD (policy ACCEPT)
> target     prot opt source               destination
>
> Chain OUTPUT (policy ACCEPT)
> target     prot opt source               destination
>
> Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.
>
>
> Vijay TH
>
>
> -==============================================================================
>
> *
>
> Date: Mon, 06 Aug 2012 06:35:04 -0400
> From: Alex Balashov <abalashov at evaristesys.com> <abalashov at evaristesys.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: sr-users at lists.sip-router.org
> Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com> <17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com>
> Content-Type: text/plain; charset="utf-8"
>
> You might consider pasting the actual output of: iptables -L -n
>
> This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
>
> Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> <vijay.thakur at loopmethods.com> wrote:Hi all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
>
> Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
>
> I have not implemented any firewall. You can check the out put of my
> iptables : http://pastebin.com/i3zUfVeb
>
> The SIP server is hosted at linnode.
>
> With thanks in advance. Sorry dual posting.
>
> Vijay TH
>
>
> *
> *
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From abalashov at evaristesys.com  Mon Aug  6 15:04:04 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 06 Aug 2012 09:04:04 -0400
Subject: [SR-Users] Kernel Droping SIP packet
Message-ID: <om9cr6896brukhcdg091wk1n.1344258244731@email.android.com>

Ah, you short-circuited where I was going with this. Couldn't remember the name of the module.?

Yep, Vijay. What Richard said.?




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Richard Fuchs <rfuchs at sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur wrote:
> Hi all,
> 
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
> 
> Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

This is coming from nf_conntrack_sip, which is a netfilter connection
tracking kernel module for SIP. I've never used it, but judging from
what Google brings up, it seems to be very buggy. You should be able to
just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
and/or if you want to keep it from auto-loading, you can blacklist it in
/etc/modprobe.d/ and then reboot.

HTH


_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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From graham at g-rock.net  Sun Aug  5 17:33:22 2012
From: graham at g-rock.net (Graham Wooden (personal))
Date: Sun, 05 Aug 2012 10:33:22 -0500
Subject: [SR-Users] LCR - inconsistent return results for gateway selection.
Message-ID: <CC43FC72.1B0FA%graham@g-rock.net>

Hello,

I am noticing that some of the time (not always consistent) that load_gw()
/ next_gw() will not always return the gateway with the lowest priority
number. 

For example, when I look up a route for 205595, I have three grp_ids
(gateways) that services the destination; with them being separated with
1, 2, and 3 priorities:

id      prefix  from_uri        grp_id  priority
7468    205595  NULL    	1       1
7469    205595  NULL    	2       2
7470    205595  NULL    	7       3

About 1/2 of the time, it will chose the priority 1 gateway, as it should.
But randomly, it will choose the other gateways on the first try.  In this
particular case, all three gateways are going to the same destination IP
(just using different hostnames in the "gw" mySQL table) - so I know the
gateways are up. Also, I am not noticing any failures and going to the
next gateway, like it choosing #1, it failing and then going to #2.

What kind of debugging can I do on this gateway selection process? I am
running 1.5.5 notls, and have approximately 425K routes. Kamailio is
started with "-m 512" for the additional memory.

Thanks,

-graham




From jh at tutpro.com  Mon Aug  6 19:12:59 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Mon, 6 Aug 2012 20:12:59 +0300
Subject: [SR-Users] LCR - inconsistent return results for gateway selection.
In-Reply-To: <CC43FC72.1B0FA%graham@g-rock.net>
References: <CC43FC72.1B0FA%graham@g-rock.net>
Message-ID: <20511.64283.741998.427448@siika.tutpro.com>

Graham Wooden (personal) writes:

> What kind of debugging can I do on this gateway selection process? I am
> running 1.5.5 notls, and have approximately 425K routes. Kamailio is
> started with "-m 512" for the additional memory.

graham,

1.5.5 is quite old and lcr implementation has changed a lot since then.
i would suggest upgrading.

-- juha


From graham at g-rock.net  Mon Aug  6 20:14:46 2012
From: graham at g-rock.net (Graham Wooden (personal))
Date: Mon, 06 Aug 2012 13:14:46 -0500
Subject: [SR-Users] LCR - inconsistent return results for gateway
 selection.
In-Reply-To: <20511.64283.741998.427448@siika.tutpro.com>
Message-ID: <CC457360.1B3AC%graham@g-rock.net>

Thanks for the reply Juha.  I have a 3.3.0 system that I have been trying
to get my routing logic converted over. I will keep working on that.
Thanks again,

-graham


On 8/6/12 12:12 PM, "Juha Heinanen" <jh at tutpro.com> wrote:

>Graham Wooden (personal) writes:
>
>> What kind of debugging can I do on this gateway selection process? I am
>> running 1.5.5 notls, and have approximately 425K routes. Kamailio is
>> started with "-m 512" for the additional memory.
>
>graham,
>
>1.5.5 is quite old and lcr implementation has changed a lot since then.
>i would suggest upgrading.
>
>-- juha
>
>_______________________________________________
>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>sr-users at lists.sip-router.org
>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




From jh at tutpro.com  Mon Aug  6 20:36:53 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Mon, 6 Aug 2012 21:36:53 +0300
Subject: [SR-Users] LCR - inconsistent return results for gateway
 selection.
In-Reply-To: <CC457360.1B3AC%graham@g-rock.net>
References: <20511.64283.741998.427448@siika.tutpro.com>
	<CC457360.1B3AC%graham@g-rock.net>
Message-ID: <20512.3781.811124.225056@siika.tutpro.com>

Graham Wooden (personal) writes:

> Thanks for the reply Juha.  I have a 3.3.0 system that I have been trying
> to get my routing logic converted over. I will keep working on that.

ok, let the list know if you experience any lcr problems with 3.3 and
we'll try to fix them.

-- juha


From brandon at cryy.com  Tue Aug  7 00:27:11 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Mon, 6 Aug 2012 15:27:11 -0700
Subject: [SR-Users] AVPOPS / TM behavior
Message-ID: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>

Hello,

   I am curious if there is any documentation on how AVP's processing works
in the following scenario below.

UAC 1 -> KAMAILIO -> KAMAILIO -> DEST

It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I relay
back to the same KAMAILIO proxy (self)?

Is there any documentation on why or when this would occur?

Is there a better way to handle such a scenario?  i.e. more dynamic
internal routing, vs relaying to self.

Thanks as always in advance!

Sincerely,
Brandon Armstead
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From vijay.thakur at loopmethods.com  Tue Aug  7 06:54:41 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Tue, 07 Aug 2012 10:24:41 +0530
Subject: [SR-Users] sr-users Digest, Vol 87, Issue 18
In-Reply-To: <mailman.4205.1344265016.2421.sr-users@lists.sip-router.org>
References: <mailman.4205.1344265016.2421.sr-users@lists.sip-router.org>
Message-ID: <50209F91.70200@loopmethods.com>

Thanks all of your for reply. Here is the output of command:


root at li496-23:~# lsmod | grep -i sip

root at li496-23:~#

This command is giving blank output.

This is linnode server. So the zcat /proc/config.gz command reveals the 
kernel loaded modules.
You can see complete output at pastebin:

http://pastebin.com/a0iqpc4D.

Thanks in advance

Vijay Th.





On Monday 06 August 2012 08:26 PM, sr-users-request at lists.sip-router.org 
wrote:
> Send sr-users mailing list submissions to
> 	sr-users at lists.sip-router.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> or, via email, send a message with subject or body 'help' to
> 	sr-users-request at lists.sip-router.org
>
> You can reach the person managing the list at
> 	sr-users-owner at lists.sip-router.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of sr-users digest..."
>
>
> Today's Topics:
>
>     1. Re: Kernel Droping SIP packet (Richard Fuchs)
>     2. Re: Kernel Droping SIP packet (Alex Balashov)
>     3. Re: Kernel Droping SIP packet (Konstantin M.)
>     4. Re: Kernel Droping SIP packet (Alex Balashov)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 06 Aug 2012 09:01:03 -0400
> From: Richard Fuchs <rfuchs at sipwise.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: sr-users at lists.sip-router.org
> Message-ID: <501FC00F.6000604 at sipwise.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On 08/06/12 06:08, Vijay Thakur wrote:
>> Hi all,
>>
>> I have configure Kamailio 3.1.5 Server. All things are working fine.
>> When i make a call from Soft phone (X-Lite) to iphone, all is working
>> fine. But in other case call from iphone to Softphone is not working,
>> even not ringing. During checking the logs i am getting the error:
>>
>> Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
>> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
>> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
>> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
>> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
> This is coming from nf_conntrack_sip, which is a netfilter connection
> tracking kernel module for SIP. I've never used it, but judging from
> what Google brings up, it seems to be very buggy. You should be able to
> just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
> and/or if you want to keep it from auto-loading, you can blacklist it in
> /etc/modprobe.d/ and then reboot.
>
> HTH
>
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>
> ------------------------------
>
> Message: 2
> Date: Mon, 06 Aug 2012 08:42:57 -0400
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: sr-users at lists.sip-router.org
> Message-ID: <turqvr4fgiqrutc5pk4nan43.1344256977848 at email.android.com>
> Content-Type: text/plain; charset="utf-8"
>
> Can you paste this output too??
>
> lsmod | grep -i sip
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> wrote:Thanks for prompt reply.? Here is the out put if command iptables -L -n:
>
> Chain INPUT (policy ACCEPT)
> target???? prot opt source?????????????? destination????????
>
> Chain FORWARD (policy ACCEPT)
> target???? prot opt source?????????????? destination????????
>
> Chain OUTPUT (policy ACCEPT)
> target???? prot opt source?????????????? destination??
>
> Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.
>
>
> Vijay TH
>
> -==============================================================================
>
>   Date: Mon, 06 Aug 2012 06:35:04 -0400
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: sr-users at lists.sip-router.org
> Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com>
> Content-Type: text/plain; charset="utf-8"
>
> You might consider pasting the actual output of: iptables -L -n
>
> This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> wrote:Hi all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
>
> Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
>
> I have not implemented any firewall. You can check the out put of my
> iptables : http://pastebin.com/i3zUfVeb
> The SIP server is hosted at linnode.
>
> With thanks in advance. Sorry dual posting.
>
> Vijay TH
>
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120806/756c81eb/attachment-0001.htm>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 6 Aug 2012 16:53:36 +0300
> From: "Konstantin M." <evilzluk at gmail.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> 	Users	Mailing List" <sr-users at lists.sip-router.org>
> Message-ID:
> 	<CAEYtTzeRjRK0Lks+7TWnTMP_5R5vn708Sv+um+U3VLbgj59ROw at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> rmmod nf_conntrack_sip
>
> 2012/8/6 Alex Balashov <abalashov at evaristesys.com>
>
>> Can you paste this output too?
>>
>> lsmod | grep -i sip
>>
>>
>>
>>
>> -- Alex
>>
>> --
>> Sent from my Samsung mobile, and thus lacking in the refinement one might
>> expect from a proper keyboard.
>>
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 235 E Ponce de Leon Ave
>> Suite 106
>> Decatur, GA 30030
>> Tel: +1-678-954-0670
>> Web: http://www.evaristesys.com/
>>
>> Vijay Thakur <vijay.thakur at loopmethods.com> wrote:
>> Thanks for prompt reply.  Here is the out put if command iptables -L -n:
>>
>> *Chain INPUT (policy ACCEPT)
>> target     prot opt source               destination
>>
>> Chain FORWARD (policy ACCEPT)
>> target     prot opt source               destination
>>
>> Chain OUTPUT (policy ACCEPT)
>> target     prot opt source               destination
>>
>> Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server.
>>
>>
>> Vijay TH
>>
>>
>> -==============================================================================
>>
>> *
>>
>> Date: Mon, 06 Aug 2012 06:35:04 -0400
>> From: Alex Balashov <abalashov at evaristesys.com> <abalashov at evaristesys.com>
>> Subject: Re: [SR-Users] Kernel Droping SIP packet
>> To: sr-users at lists.sip-router.org
>> Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com> <17t1fd0fbly84yy2vuy8qlcp.1344249304618 at email.android.com>
>> Content-Type: text/plain; charset="utf-8"
>>
>> You might consider pasting the actual output of: iptables -L -n
>>
>> This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.?
>>
>>
>>
>>
>> -- Alex
>>
>> --
>> Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
>>
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 235 E Ponce de Leon Ave
>> Suite 106
>> Decatur, GA 30030
>> Tel: +1-678-954-0670
>>
>> Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur at loopmethods.com> <vijay.thakur at loopmethods.com> wrote:Hi all,
>>
>> I have configure Kamailio 3.1.5 Server. All things are working fine.
>> When i make a call from Soft phone (X-Lite) to iphone, all is working
>> fine. But in other case call from iphone to Softphone is not working,
>> even not ringing. During checking the logs i am getting the error:
>>
>> Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
>> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
>> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
>> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
>> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
>>
>> I have not implemented any firewall. You can check the out put of my
>> iptables : http://pastebin.com/i3zUfVeb
>>
>> The SIP server is hosted at linnode.
>>
>> With thanks in advance. Sorry dual posting.
>>
>> Vijay TH
>>
>>
>> *
>> *
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120806/2552ea53/attachment-0001.htm>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 06 Aug 2012 09:04:04 -0400
> From: Alex Balashov <abalashov at evaristesys.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: sr-users at lists.sip-router.org
> Message-ID: <om9cr6896brukhcdg091wk1n.1344258244731 at email.android.com>
> Content-Type: text/plain; charset="utf-8"
>
> Ah, you short-circuited where I was going with this. Couldn't remember the name of the module.?
>
> Yep, Vijay. What Richard said.?
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/Richard Fuchs <rfuchs at sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur wrote:
>> Hi all,
>>
>> I have configure Kamailio 3.1.5 Server. All things are working fine.
>> When i make a call from Soft phone (X-Lite) to iphone, all is working
>> fine. But in other case call from iphone to Softphone is not working,
>> even not ringing. During checking the logs i am getting the error:
>>
>> Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
>> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
>> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
>> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
>> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
> This is coming from nf_conntrack_sip, which is a netfilter connection
> tracking kernel module for SIP. I've never used it, but judging from
> what Google brings up, it seems to be very buggy. You should be able to
> just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
> and/or if you want to keep it from auto-loading, you can blacklist it in
> /etc/modprobe.d/ and then reboot.
>
> HTH
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> End of sr-users Digest, Vol 87, Issue 18
> ****************************************


-- 
Best Regards,

Vijay Thakur
(Assistant Manager - Networks)
Mobile   : +91 8744018065
Mail     : vijay.thakur at loopmethods.com

Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web: www.loopmethods.com

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From klaus.mailinglists at pernau.at  Tue Aug  7 11:32:44 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 07 Aug 2012 11:32:44 +0200
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
Message-ID: <5020E0BC.4050705@pernau.at>

AVPs are associated with the transaction. If you "spiral" a request 
through the same proxy, then for the proxy it is a new transaction. 
Thus, when processing the request a second time, there is a new 
transaction and you do not have access to the AVPs of the previous 
transaction.

Workarounds are:
- store data in SIP headers and retrieve it later (ugly)
- use htable module to store data during transaction 1 and retrieve it 
during transaction 2. Therefore you need a known "key" which is 
identical in this 2 transactions only (e.g. use "$ci$ft" as base for the 
key).

regards
Klaus




On 07.08.2012 00:27, Brandon Armstead wrote:
> Hello,
>
>     I am curious if there is any documentation on how AVP's processing
> works in the following scenario below.
>
> UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>
> It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
> relay back to the same KAMAILIO proxy (self)?
>
> Is there any documentation on why or when this would occur?
>
> Is there a better way to handle such a scenario?  i.e. more dynamic
> internal routing, vs relaying to self.
>
> Thanks as always in advance!
>
> Sincerely,
> Brandon Armstead
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From klaus.mailinglists at pernau.at  Tue Aug  7 11:32:44 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 07 Aug 2012 11:32:44 +0200
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
Message-ID: <5020E0BC.4050705@pernau.at>

AVPs are associated with the transaction. If you "spiral" a request 
through the same proxy, then for the proxy it is a new transaction. 
Thus, when processing the request a second time, there is a new 
transaction and you do not have access to the AVPs of the previous 
transaction.

Workarounds are:
- store data in SIP headers and retrieve it later (ugly)
- use htable module to store data during transaction 1 and retrieve it 
during transaction 2. Therefore you need a known "key" which is 
identical in this 2 transactions only (e.g. use "$ci$ft" as base for the 
key).

regards
Klaus




On 07.08.2012 00:27, Brandon Armstead wrote:
> Hello,
>
>     I am curious if there is any documentation on how AVP's processing
> works in the following scenario below.
>
> UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>
> It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
> relay back to the same KAMAILIO proxy (self)?
>
> Is there any documentation on why or when this would occur?
>
> Is there a better way to handle such a scenario?  i.e. more dynamic
> internal routing, vs relaying to self.
>
> Thanks as always in advance!
>
> Sincerely,
> Brandon Armstead
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From brandon at cryy.com  Tue Aug  7 18:22:24 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 7 Aug 2012 09:22:24 -0700
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <5020E0BC.4050705@pernau.at>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
Message-ID: <CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>

Klaus,

   Thank you for this detailed explanation.  This is essentially what I
figured was happening.  I was able to use htable to work around it.

I guess however I am still confused as to where there is any public
documentation on this specific bit.  Had I've not been working with
Kamailio for years I would think this would confuse others.

Let me know if it is somewhere else, otherwise I will add it to the
Kamailio wiki.

Sincerely,
Brandon Armstead

On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion <klaus.mailinglists at pernau.at
> wrote:

> AVPs are associated with the transaction. If you "spiral" a request
> through the same proxy, then for the proxy it is a new transaction. Thus,
> when processing the request a second time, there is a new transaction and
> you do not have access to the AVPs of the previous transaction.
>
> Workarounds are:
> - store data in SIP headers and retrieve it later (ugly)
> - use htable module to store data during transaction 1 and retrieve it
> during transaction 2. Therefore you need a known "key" which is identical
> in this 2 transactions only (e.g. use "$ci$ft" as base for the key).
>
> regards
> Klaus
>
>
>
>
>
> On 07.08.2012 00:27, Brandon Armstead wrote:
>
>> Hello,
>>
>>     I am curious if there is any documentation on how AVP's processing
>> works in the following scenario below.
>>
>> UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>>
>> It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
>> relay back to the same KAMAILIO proxy (self)?
>>
>> Is there any documentation on why or when this would occur?
>>
>> Is there a better way to handle such a scenario?  i.e. more dynamic
>> internal routing, vs relaying to self.
>>
>> Thanks as always in advance!
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> ______________________________**_________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
-------------- next part --------------
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From brandon at cryy.com  Tue Aug  7 18:22:24 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 7 Aug 2012 09:22:24 -0700
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <5020E0BC.4050705@pernau.at>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
Message-ID: <CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>

Klaus,

   Thank you for this detailed explanation.  This is essentially what I
figured was happening.  I was able to use htable to work around it.

I guess however I am still confused as to where there is any public
documentation on this specific bit.  Had I've not been working with
Kamailio for years I would think this would confuse others.

Let me know if it is somewhere else, otherwise I will add it to the
Kamailio wiki.

Sincerely,
Brandon Armstead

On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion <klaus.mailinglists at pernau.at
> wrote:

> AVPs are associated with the transaction. If you "spiral" a request
> through the same proxy, then for the proxy it is a new transaction. Thus,
> when processing the request a second time, there is a new transaction and
> you do not have access to the AVPs of the previous transaction.
>
> Workarounds are:
> - store data in SIP headers and retrieve it later (ugly)
> - use htable module to store data during transaction 1 and retrieve it
> during transaction 2. Therefore you need a known "key" which is identical
> in this 2 transactions only (e.g. use "$ci$ft" as base for the key).
>
> regards
> Klaus
>
>
>
>
>
> On 07.08.2012 00:27, Brandon Armstead wrote:
>
>> Hello,
>>
>>     I am curious if there is any documentation on how AVP's processing
>> works in the following scenario below.
>>
>> UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>>
>> It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
>> relay back to the same KAMAILIO proxy (self)?
>>
>> Is there any documentation on why or when this would occur?
>>
>> Is there a better way to handle such a scenario?  i.e. more dynamic
>> internal routing, vs relaying to self.
>>
>> Thanks as always in advance!
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> ______________________________**_________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
-------------- next part --------------
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From klaus.mailinglists at pernau.at  Tue Aug  7 21:00:29 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 07 Aug 2012 21:00:29 +0200
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
	<CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
Message-ID: <502165CD.60703@pernau.at>

Once you know it you will find it :-)

http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps

regards
Klaus

On 07.08.2012 18:22, Brandon Armstead wrote:
> Klaus,
>
>     Thank you for this detailed explanation.  This is essentially what I
> figured was happening.  I was able to use htable to work around it.
>
> I guess however I am still confused as to where there is any public
> documentation on this specific bit.  Had I've not been working with
> Kamailio for years I would think this would confuse others.
>
> Let me know if it is somewhere else, otherwise I will add it to the
> Kamailio wiki.
>
> Sincerely,
> Brandon Armstead
>
> On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
>
>     AVPs are associated with the transaction. If you "spiral" a request
>     through the same proxy, then for the proxy it is a new transaction.
>     Thus, when processing the request a second time, there is a new
>     transaction and you do not have access to the AVPs of the previous
>     transaction.
>
>     Workarounds are:
>     - store data in SIP headers and retrieve it later (ugly)
>     - use htable module to store data during transaction 1 and retrieve
>     it during transaction 2. Therefore you need a known "key" which is
>     identical in this 2 transactions only (e.g. use "$ci$ft" as base for
>     the key).
>
>     regards
>     Klaus
>
>
>
>
>
>     On 07.08.2012 00:27, Brandon Armstead wrote:
>
>         Hello,
>
>              I am curious if there is any documentation on how AVP's
>         processing
>         works in the following scenario below.
>
>         UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>
>         It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
>         relay back to the same KAMAILIO proxy (self)?
>
>         Is there any documentation on why or when this would occur?
>
>         Is there a better way to handle such a scenario?  i.e. more dynamic
>         internal routing, vs relaying to self.
>
>         Thanks as always in advance!
>
>         Sincerely,
>         Brandon Armstead
>
>
>         _________________________________________________
>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>         mailing list
>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>         http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
>


From klaus.mailinglists at pernau.at  Tue Aug  7 21:00:29 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 07 Aug 2012 21:00:29 +0200
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
	<CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
Message-ID: <502165CD.60703@pernau.at>

Once you know it you will find it :-)

http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps

regards
Klaus

On 07.08.2012 18:22, Brandon Armstead wrote:
> Klaus,
>
>     Thank you for this detailed explanation.  This is essentially what I
> figured was happening.  I was able to use htable to work around it.
>
> I guess however I am still confused as to where there is any public
> documentation on this specific bit.  Had I've not been working with
> Kamailio for years I would think this would confuse others.
>
> Let me know if it is somewhere else, otherwise I will add it to the
> Kamailio wiki.
>
> Sincerely,
> Brandon Armstead
>
> On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
>
>     AVPs are associated with the transaction. If you "spiral" a request
>     through the same proxy, then for the proxy it is a new transaction.
>     Thus, when processing the request a second time, there is a new
>     transaction and you do not have access to the AVPs of the previous
>     transaction.
>
>     Workarounds are:
>     - store data in SIP headers and retrieve it later (ugly)
>     - use htable module to store data during transaction 1 and retrieve
>     it during transaction 2. Therefore you need a known "key" which is
>     identical in this 2 transactions only (e.g. use "$ci$ft" as base for
>     the key).
>
>     regards
>     Klaus
>
>
>
>
>
>     On 07.08.2012 00:27, Brandon Armstead wrote:
>
>         Hello,
>
>              I am curious if there is any documentation on how AVP's
>         processing
>         works in the following scenario below.
>
>         UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>
>         It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
>         relay back to the same KAMAILIO proxy (self)?
>
>         Is there any documentation on why or when this would occur?
>
>         Is there a better way to handle such a scenario?  i.e. more dynamic
>         internal routing, vs relaying to self.
>
>         Thanks as always in advance!
>
>         Sincerely,
>         Brandon Armstead
>
>
>         _________________________________________________
>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>         mailing list
>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>         http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
>


From brandon at cryy.com  Tue Aug  7 21:49:25 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 7 Aug 2012 12:49:25 -0700
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <502165CD.60703@pernau.at>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
	<CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
	<502165CD.60703@pernau.at>
Message-ID: <CACr1DbAYCuoAiYH39uJUjrY07SfZ-_rQdoFgYWWon9ALyUVcjw@mail.gmail.com>

Klaus,

   I see the following:

AVPs are special variables that are attached to SIP transactions. It is a
list of pairs (name,value). Before the transaction is created, the AVP list
is attached to SIP request. Note that the AVP list works like a stack, last
added value is retrieved first, and there can be many values for same AVP
name, an assignment to the same AVP name does not overwrite old value, it
will add the new value in the list.
While this does *technically* describe the behavior - we may want to
explicitly point out this behavior when spiraling to the same proxy.  I
guess to me its not clear enough based off of this above copied text.
 Unless I am still missing the explanation somewhere else in the text?
 Thanks!

Sincerely,
Brandon Armstead
On Tue, Aug 7, 2012 at 12:00 PM, Klaus Darilion <
klaus.mailinglists at pernau.at> wrote:

> Once you know it you will find it :-)
>
> http://www.kamailio.org/wiki/**cookbooks/3.3.x/**pseudovariables#avps<http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps>
>
> regards
> Klaus
>
>
> On 07.08.2012 18:22, Brandon Armstead wrote:
>
>> Klaus,
>>
>>     Thank you for this detailed explanation.  This is essentially what I
>> figured was happening.  I was able to use htable to work around it.
>>
>> I guess however I am still confused as to where there is any public
>> documentation on this specific bit.  Had I've not been working with
>> Kamailio for years I would think this would confuse others.
>>
>> Let me know if it is somewhere else, otherwise I will add it to the
>> Kamailio wiki.
>>
>> Sincerely,
>> Brandon Armstead
>>
>> On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
>> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists@**pernau.at<klaus.mailinglists at pernau.at>>>
>> wrote:
>>
>>     AVPs are associated with the transaction. If you "spiral" a request
>>     through the same proxy, then for the proxy it is a new transaction.
>>     Thus, when processing the request a second time, there is a new
>>     transaction and you do not have access to the AVPs of the previous
>>     transaction.
>>
>>     Workarounds are:
>>     - store data in SIP headers and retrieve it later (ugly)
>>     - use htable module to store data during transaction 1 and retrieve
>>     it during transaction 2. Therefore you need a known "key" which is
>>     identical in this 2 transactions only (e.g. use "$ci$ft" as base for
>>     the key).
>>
>>     regards
>>     Klaus
>>
>>
>>
>>
>>
>>     On 07.08.2012 00:27, Brandon Armstead wrote:
>>
>>         Hello,
>>
>>              I am curious if there is any documentation on how AVP's
>>         processing
>>         works in the following scenario below.
>>
>>         UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>>
>>         It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once
>> I
>>         relay back to the same KAMAILIO proxy (self)?
>>
>>         Is there any documentation on why or when this would occur?
>>
>>         Is there a better way to handle such a scenario?  i.e. more
>> dynamic
>>         internal routing, vs relaying to self.
>>
>>         Thanks as always in advance!
>>
>>         Sincerely,
>>         Brandon Armstead
>>
>>
>>         ______________________________**___________________
>>
>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>         mailing list
>>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-**
>> router.org <sr-users at lists.sip-router.org>>
>>         http://lists.sip-router.org/__**cgi-bin/mailman/listinfo/sr-__**
>> users <http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users><
>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>> >
>>
>>
>>
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From brandon at cryy.com  Tue Aug  7 21:49:25 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 7 Aug 2012 12:49:25 -0700
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <502165CD.60703@pernau.at>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
	<CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
	<502165CD.60703@pernau.at>
Message-ID: <CACr1DbAYCuoAiYH39uJUjrY07SfZ-_rQdoFgYWWon9ALyUVcjw@mail.gmail.com>

Klaus,

   I see the following:

AVPs are special variables that are attached to SIP transactions. It is a
list of pairs (name,value). Before the transaction is created, the AVP list
is attached to SIP request. Note that the AVP list works like a stack, last
added value is retrieved first, and there can be many values for same AVP
name, an assignment to the same AVP name does not overwrite old value, it
will add the new value in the list.
While this does *technically* describe the behavior - we may want to
explicitly point out this behavior when spiraling to the same proxy.  I
guess to me its not clear enough based off of this above copied text.
 Unless I am still missing the explanation somewhere else in the text?
 Thanks!

Sincerely,
Brandon Armstead
On Tue, Aug 7, 2012 at 12:00 PM, Klaus Darilion <
klaus.mailinglists at pernau.at> wrote:

> Once you know it you will find it :-)
>
> http://www.kamailio.org/wiki/**cookbooks/3.3.x/**pseudovariables#avps<http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps>
>
> regards
> Klaus
>
>
> On 07.08.2012 18:22, Brandon Armstead wrote:
>
>> Klaus,
>>
>>     Thank you for this detailed explanation.  This is essentially what I
>> figured was happening.  I was able to use htable to work around it.
>>
>> I guess however I am still confused as to where there is any public
>> documentation on this specific bit.  Had I've not been working with
>> Kamailio for years I would think this would confuse others.
>>
>> Let me know if it is somewhere else, otherwise I will add it to the
>> Kamailio wiki.
>>
>> Sincerely,
>> Brandon Armstead
>>
>> On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
>> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists@**pernau.at<klaus.mailinglists at pernau.at>>>
>> wrote:
>>
>>     AVPs are associated with the transaction. If you "spiral" a request
>>     through the same proxy, then for the proxy it is a new transaction.
>>     Thus, when processing the request a second time, there is a new
>>     transaction and you do not have access to the AVPs of the previous
>>     transaction.
>>
>>     Workarounds are:
>>     - store data in SIP headers and retrieve it later (ugly)
>>     - use htable module to store data during transaction 1 and retrieve
>>     it during transaction 2. Therefore you need a known "key" which is
>>     identical in this 2 transactions only (e.g. use "$ci$ft" as base for
>>     the key).
>>
>>     regards
>>     Klaus
>>
>>
>>
>>
>>
>>     On 07.08.2012 00:27, Brandon Armstead wrote:
>>
>>         Hello,
>>
>>              I am curious if there is any documentation on how AVP's
>>         processing
>>         works in the following scenario below.
>>
>>         UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>>
>>         It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once
>> I
>>         relay back to the same KAMAILIO proxy (self)?
>>
>>         Is there any documentation on why or when this would occur?
>>
>>         Is there a better way to handle such a scenario?  i.e. more
>> dynamic
>>         internal routing, vs relaying to self.
>>
>>         Thanks as always in advance!
>>
>>         Sincerely,
>>         Brandon Armstead
>>
>>
>>         ______________________________**___________________
>>
>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>         mailing list
>>         sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-**
>> router.org <sr-users at lists.sip-router.org>>
>>         http://lists.sip-router.org/__**cgi-bin/mailman/listinfo/sr-__**
>> users <http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users><
>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>> >
>>
>>
>>
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From nboric at yx.cl  Tue Aug  7 22:34:43 2012
From: nboric at yx.cl (Neven Boric)
Date: Tue, 07 Aug 2012 16:34:43 -0400
Subject: [SR-Users] set_advertised_port using pseudovariable
Message-ID: <50217BE3.9000801@yx.cl>

Hi,

I'm trying to use Kamailio as an outbound proxy behind a NAT (all 
clients and kamailio itself are behind the same NAT):

UACs -> Kamailio -> NAT router -> PBX (hosted, public server)

I figured I could detect the external source port used by the router by 
periodically sending an OPTIONS request to the public server, then 
capture the rport value in the reply, and use that port as an input to 
set_advertised_port. Because I'm doing this periodically, it should have 
the added value of keeping the NAT mapping alive. Otherwise, I should be 
able to detect it, as the rport value will change.

I managed to periodically send the OPTIONS request, read the rport value 
on the reply message and store it on a pseudovariable using:

$var(rport) = $sel(v.rport);

But when I try to use that value to set the advertised port, I get an 
error, as if set_advertised_port only accepted literal values:

if( $var(rport) != 0)
{
         set_advertised_port($var(rport));
}

Not starting Kamailio: invalid configuration file!
-e
  0(4692) : <core> [cfg.y:3591]: parse error in config file 
/usr/local/etc/kamailio/kamailio.cfg, line 504, column 24-34: syntax error
  0(4692) : <core> [cfg.y:3594]: parse error in config file 
/usr/local/etc/kamailio/kamailio.cfg, line 504, column 35: bad argument, 
string expected
ERROR: bad config file (2 errors)

So, am I missing something or does set_advertised_port actually only 
accept literal values?

Thanks
Neven Boric


From abalashov at evaristesys.com  Tue Aug  7 22:38:16 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Tue, 07 Aug 2012 16:38:16 -0400
Subject: [SR-Users] set_advertised_port using pseudovariable
In-Reply-To: <50217BE3.9000801@yx.cl>
References: <50217BE3.9000801@yx.cl>
Message-ID: <50217CB8.30507@evaristesys.com>

On 08/07/2012 04:34 PM, Neven Boric wrote:

> So, am I missing something or does set_advertised_port actually only
> accept literal values?

Correct, set_advertised_port does not accept PVs.  There is an entire 
category of legacy core functions, and some module functions, for which 
that is true.

There's an ongoing effort to update many of them to use PVs, but it's 
... incremental.  :-)

Methodologically speaking, you really, really don't want to run Kamailio 
behind a NAT as a client, unless the endpoint on the other side of the 
NAT can deal with it entirely using far-end NAT traversal strategies.

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From nboric at yx.cl  Tue Aug  7 23:23:50 2012
From: nboric at yx.cl (Neven Boric)
Date: Tue, 07 Aug 2012 17:23:50 -0400
Subject: [SR-Users] set_advertised_port using pseudovariable
In-Reply-To: <50217CB8.30507@evaristesys.com>
References: <50217BE3.9000801@yx.cl> <50217CB8.30507@evaristesys.com>
Message-ID: <50218766.60204@yx.cl>


Alex Balashov escribi?:
> On 08/07/2012 04:34 PM, Neven Boric wrote:
>
>> So, am I missing something or does set_advertised_port actually only
>> accept literal values?
>
> Correct, set_advertised_port does not accept PVs.  There is an entire 
> category of legacy core functions, and some module functions, for 
> which that is true.
>
> There's an ongoing effort to update many of them to use PVs, but it's 
> ... incremental.  :-)
I wonder how hard would it be to do it myself. Just getting to know 
Kamailio as a user, haven't looked at the code yet. Any pointers? 
Hopefully some other function where this was recently implemented, so I 
can browse the commits.
>
> Methodologically speaking, you really, really don't want to run 
> Kamailio behind a NAT as a client, unless the endpoint on the other 
> side of the NAT can deal with it entirely using far-end NAT traversal 
> strategies.
There's this trend here (and I assume elsewhere) of providers offering 
hosted (i.e remote) PBX and installing an "SBC" on customer premises to 
access said PBX. I use quotes for SBC, because, from what I've gathered, 
it can mean different things to different people. In this case, it's a 
device that can be used to access the remote PBX without worring about 
NAT issues, handles security, does SIP normalization, topology hiding, 
maybe encryption, etc. One important feature is survivability, as in 
UACs being able to talk to each other (and maybe do conferences and 
other media stuff, so you also need a media server) in case the remote 
PBX is temporarily inaccessible. I just assumed they are installed 
behind the NAT, because all manufacturers (AudioCodes, Mediatrix and the 
like) mention NAT traversal as an important feature. Also, most of the 
times the clients are SMBs without access (or lacking the knowledge, or 
just hesitant because of security reasons) to configure port forwarding 
on their router.

Are you familiar with this scenario? Would you say I'm wrong in thinking 
these devices are installed behind the NAT? Or that Kamailio is not a 
good choice for this scenario?

Thanks
--
Neven Boric


From ngoahotanglongbk at gmail.com  Wed Aug  8 07:59:46 2012
From: ngoahotanglongbk at gmail.com (Duong Manh Truong)
Date: Wed, 8 Aug 2012 12:59:46 +0700
Subject: [SR-Users] Kamailio with APNS
Message-ID: <CADNPFHRR_nu4RNZXjLLsJtHmvNzxdcSY6pRkRx+exzwoucmy1A@mail.gmail.com>

As your first solution with UAC module:

"As a plan B we instead opted for sending via uac module the invite to an
oversip instance (it's open-source since a week or two), which triggers
an HTTP request towards APNS, and on kamailio check with inv-timers
every few seconds whether the client came online in the meanwhile, then
complete the call. The drawback is that you can wait only for so many
loop interations due to the max branch limitation in kamailio (you'd
need to recompile it to set it higher)."

I read about the UAC module but feeling very hard to implement it as
your suggestion,

http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html

Please give me and others more details about this?

Which functions those we need ?

Are they in: UAC module, Exec module, tm module ??

Thanks
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From Marco.Barthel at de.bosch.com  Wed Aug  8 09:34:21 2012
From: Marco.Barthel at de.bosch.com (Barthel Marco (CI/AFU1))
Date: Wed, 8 Aug 2012 09:34:21 +0200
Subject: [SR-Users] local_timer: add_timeout: 0 expire timer added
Message-ID: <44C56CFBDEDE184893B6FAD3401140D141254E71A3@SI-MBX02.de.bosch.com>

Hi,
 Occasionally I can see the following log output from Kamailio in our systems:

WARNING: <core> [local_timer.c:97]: WARNING: local_timer: add_timeout: 0 expire timer added

Can someone explain what it means?

Many thanks.

Mit freundlichen Gr??en / Best regards

Marco Barthel

Robert Bosch GmbH
 (CI/AFU1)
www.bosch.com<http://www.bosch.com>

Tel. +49 711 811-3602341
marco.barthel at de.bosch.com<mailto:marco.barthel at de.bosch.com>

Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000;
Aufsichtsratsvorsitzender: Franz Fehrenbach; Gesch?ftsf?hrung: Volkmar Denner, Siegfried Dais;
Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Dirk Hoheisel, Christoph K?bel, Uwe Raschke,
Wolf-Henning Scheider, Werner Struth, Peter Tyroller



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From klaus.mailinglists at pernau.at  Wed Aug  8 10:02:33 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 08 Aug 2012 10:02:33 +0200
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <CACr1DbAYCuoAiYH39uJUjrY07SfZ-_rQdoFgYWWon9ALyUVcjw@mail.gmail.com>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
	<CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
	<502165CD.60703@pernau.at>
	<CACr1DbAYCuoAiYH39uJUjrY07SfZ-_rQdoFgYWWon9ALyUVcjw@mail.gmail.com>
Message-ID: <50221D19.9000305@pernau.at>

The description is implicit: "AVPs are special variables that are 
attached to SIP transactions." As a spiral causes 2 transactions there 
are two different contexts for AVPs.

Feel free to improve/extend the description.

regards
Klaus

On 07.08.2012 21:49, Brandon Armstead wrote:
> Klaus,
>
>     I see the following:
>
> AVPs are special variables that are attached to SIP transactions. It is
> a list of pairs (name,value). Before the transaction is created, the AVP
> list is attached to SIP request. Note that the AVP list works like a
> stack, last added value is retrieved first, and there can be many values
> for same AVP name, an assignment to the same AVP name does not overwrite
> old value, it will add the new value in the list.
>
> While this does *technically* describe the behavior - we may want to
> explicitly point out this behavior when spiraling to the same proxy.  I
> guess to me its not clear enough based off of this above copied text.
>   Unless I am still missing the explanation somewhere else in the text?
>   Thanks!
>
> Sincerely,
> Brandon Armstead
> On Tue, Aug 7, 2012 at 12:00 PM, Klaus Darilion
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
>
>     Once you know it you will find it :-)
>
>     http://www.kamailio.org/wiki/__cookbooks/3.3.x/__pseudovariables#avps <http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps>
>
>     regards
>     Klaus
>
>
>     On 07.08.2012 18:22, Brandon Armstead wrote:
>
>         Klaus,
>
>              Thank you for this detailed explanation.  This is
>         essentially what I
>         figured was happening.  I was able to use htable to work around it.
>
>         I guess however I am still confused as to where there is any public
>         documentation on this specific bit.  Had I've not been working with
>         Kamailio for years I would think this would confuse others.
>
>         Let me know if it is somewhere else, otherwise I will add it to the
>         Kamailio wiki.
>
>         Sincerely,
>         Brandon Armstead
>
>         On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
>         <klaus.mailinglists at pernau.at
>         <mailto:klaus.mailinglists at pernau.at>
>         <mailto:klaus.mailinglists at __pernau.at
>         <mailto:klaus.mailinglists at pernau.at>>> wrote:
>
>              AVPs are associated with the transaction. If you "spiral" a
>         request
>              through the same proxy, then for the proxy it is a new
>         transaction.
>              Thus, when processing the request a second time, there is a new
>              transaction and you do not have access to the AVPs of the
>         previous
>              transaction.
>
>              Workarounds are:
>              - store data in SIP headers and retrieve it later (ugly)
>              - use htable module to store data during transaction 1 and
>         retrieve
>              it during transaction 2. Therefore you need a known "key"
>         which is
>              identical in this 2 transactions only (e.g. use "$ci$ft" as
>         base for
>              the key).
>
>              regards
>              Klaus
>
>
>
>
>
>              On 07.08.2012 00:27, Brandon Armstead wrote:
>
>                  Hello,
>
>                       I am curious if there is any documentation on how
>         AVP's
>                  processing
>                  works in the following scenario below.
>
>                  UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>
>                  It seems that AVP's I set between UAC 1 -> KAMAILIO are
>         lost once I
>                  relay back to the same KAMAILIO proxy (self)?
>
>                  Is there any documentation on why or when this would occur?
>
>                  Is there a better way to handle such a scenario?  i.e.
>         more dynamic
>                  internal routing, vs relaying to self.
>
>                  Thanks as always in advance!
>
>                  Sincerely,
>                  Brandon Armstead
>
>
>                  ___________________________________________________
>
>                  SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>                  mailing list
>         sr-users at lists.sip-router.org
>         <mailto:sr-users at lists.sip-router.org>
>         <mailto:sr-users at lists.sip-__router.org
>         <mailto:sr-users at lists.sip-router.org>>
>         http://lists.sip-router.org/____cgi-bin/mailman/listinfo/sr-____users
>         <http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users>
>         <http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users
>         <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>
>
>
>


From klaus.mailinglists at pernau.at  Wed Aug  8 10:02:33 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 08 Aug 2012 10:02:33 +0200
Subject: [SR-Users] AVPOPS / TM behavior
In-Reply-To: <CACr1DbAYCuoAiYH39uJUjrY07SfZ-_rQdoFgYWWon9ALyUVcjw@mail.gmail.com>
References: <CACr1DbD4vR_Qu0BbNMv9GGn3ewDiNVsunWC+h=LuZ50khp=x9w@mail.gmail.com>
	<5020E0BC.4050705@pernau.at>
	<CACr1DbBgS2ELuXm16Lr7KZf+HSd5_fgv6HEtHEts2dyYuDx=_w@mail.gmail.com>
	<502165CD.60703@pernau.at>
	<CACr1DbAYCuoAiYH39uJUjrY07SfZ-_rQdoFgYWWon9ALyUVcjw@mail.gmail.com>
Message-ID: <50221D19.9000305@pernau.at>

The description is implicit: "AVPs are special variables that are 
attached to SIP transactions." As a spiral causes 2 transactions there 
are two different contexts for AVPs.

Feel free to improve/extend the description.

regards
Klaus

On 07.08.2012 21:49, Brandon Armstead wrote:
> Klaus,
>
>     I see the following:
>
> AVPs are special variables that are attached to SIP transactions. It is
> a list of pairs (name,value). Before the transaction is created, the AVP
> list is attached to SIP request. Note that the AVP list works like a
> stack, last added value is retrieved first, and there can be many values
> for same AVP name, an assignment to the same AVP name does not overwrite
> old value, it will add the new value in the list.
>
> While this does *technically* describe the behavior - we may want to
> explicitly point out this behavior when spiraling to the same proxy.  I
> guess to me its not clear enough based off of this above copied text.
>   Unless I am still missing the explanation somewhere else in the text?
>   Thanks!
>
> Sincerely,
> Brandon Armstead
> On Tue, Aug 7, 2012 at 12:00 PM, Klaus Darilion
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
>
>     Once you know it you will find it :-)
>
>     http://www.kamailio.org/wiki/__cookbooks/3.3.x/__pseudovariables#avps <http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps>
>
>     regards
>     Klaus
>
>
>     On 07.08.2012 18:22, Brandon Armstead wrote:
>
>         Klaus,
>
>              Thank you for this detailed explanation.  This is
>         essentially what I
>         figured was happening.  I was able to use htable to work around it.
>
>         I guess however I am still confused as to where there is any public
>         documentation on this specific bit.  Had I've not been working with
>         Kamailio for years I would think this would confuse others.
>
>         Let me know if it is somewhere else, otherwise I will add it to the
>         Kamailio wiki.
>
>         Sincerely,
>         Brandon Armstead
>
>         On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
>         <klaus.mailinglists at pernau.at
>         <mailto:klaus.mailinglists at pernau.at>
>         <mailto:klaus.mailinglists at __pernau.at
>         <mailto:klaus.mailinglists at pernau.at>>> wrote:
>
>              AVPs are associated with the transaction. If you "spiral" a
>         request
>              through the same proxy, then for the proxy it is a new
>         transaction.
>              Thus, when processing the request a second time, there is a new
>              transaction and you do not have access to the AVPs of the
>         previous
>              transaction.
>
>              Workarounds are:
>              - store data in SIP headers and retrieve it later (ugly)
>              - use htable module to store data during transaction 1 and
>         retrieve
>              it during transaction 2. Therefore you need a known "key"
>         which is
>              identical in this 2 transactions only (e.g. use "$ci$ft" as
>         base for
>              the key).
>
>              regards
>              Klaus
>
>
>
>
>
>              On 07.08.2012 00:27, Brandon Armstead wrote:
>
>                  Hello,
>
>                       I am curious if there is any documentation on how
>         AVP's
>                  processing
>                  works in the following scenario below.
>
>                  UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>
>                  It seems that AVP's I set between UAC 1 -> KAMAILIO are
>         lost once I
>                  relay back to the same KAMAILIO proxy (self)?
>
>                  Is there any documentation on why or when this would occur?
>
>                  Is there a better way to handle such a scenario?  i.e.
>         more dynamic
>                  internal routing, vs relaying to self.
>
>                  Thanks as always in advance!
>
>                  Sincerely,
>                  Brandon Armstead
>
>
>                  ___________________________________________________
>
>                  SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>                  mailing list
>         sr-users at lists.sip-router.org
>         <mailto:sr-users at lists.sip-router.org>
>         <mailto:sr-users at lists.sip-__router.org
>         <mailto:sr-users at lists.sip-router.org>>
>         http://lists.sip-router.org/____cgi-bin/mailman/listinfo/sr-____users
>         <http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users>
>         <http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users
>         <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>
>
>
>


From vijay.thakur at loopmethods.com  Wed Aug  8 12:53:21 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Wed, 08 Aug 2012 16:23:21 +0530
Subject: [SR-Users] Kernel Module Issue
Message-ID: <50224521.1030209@loopmethods.com>

Thanks for the hint. What is the file name in /etc/modprob.d/ where i am supposed to enter this module name to disable it.

Thanks

Vijay Tha

Message: 1
Date: Mon, 06 Aug 2012 09:01:03 -0400
From: Richard Fuchs<rfuchs at sipwise.com>
Subject: Re: [SR-Users] Kernel Droping SIP packet
To:sr-users at lists.sip-router.org
Message-ID:<501FC00F.6000604 at sipwise.com>
Content-Type: text/plain; charset="iso-8859-1"

On 08/06/12 06:08, Vijay Thakur wrote:

> Hi all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
>
> Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)

This is coming from nf_conntrack_sip, which is a netfilter connection
tracking kernel module for SIP. I've never used it, but judging from
what Google brings up, it seems to be very buggy. You should be able to
just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
and/or if you want to keep it from auto-loading, you can blacklist it in
/etc/modprobe.d/  and then reboot.

HTH

copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited.
-----------------------------------------------------------------------------------------------------------------

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From fchahrour at vanrise.com  Wed Aug  8 17:32:27 2012
From: fchahrour at vanrise.com (Fatima Chahrour~Vanrise Support)
Date: Wed, 8 Aug 2012 18:32:27 +0300
Subject: [SR-Users] Kamailio LCR Module test scenario
Message-ID: <01bd01cd757b$0543d090$0fcb71b0$@com>

Dears,

 

Am trying to apply Kamailio LCR feature lab test, after am finally able to
run Kamailio with no errors, am not able to reach my target clarified in
this following scenario:

Call 961312345 - route the call to 192.x.x.15 in first place and if failed
route the call to 192.x.x.10. I applied needed configuration in lcr tables
(attached image)  and the routing script is:

 

route{

       if (!mf_process_maxfwd_header("10")) {

               sl_send_reply("483","Too Many Hops");

               exit;

       };

       if (msg:len >= 2048 ) {

               sl_send_reply("513", "Message too big");

               exit;

       };

 

if (!method=="REGISTER")

        record_route();

 

if (loose_route()) {

        append_hf("P-hint: rr-enforced\r\n");

        route(1);

};

 

 

if (!uri==myself) {

                append_hf("P-hint: outbound\r\n");

        route(1);

};

 

if (is_method("INVITE"))

       {

              if (!load_gws(1)) {

              sl_send_reply("503", "Unable to load gateways");

              exit;

              }

}

 

if (uri==myself) {

        if (method=="REGISTER") { 

                 save("location");

                 exit;

        };

        lookup("aliases");

        if (!uri==myself) {

                 append_hf("P-hint: outbound alias\r\n");

                 route(1);

                        };

 

        # native SIP destinations are handled using ourUSRLOC DB

        if (!lookup("location")) {

                 sl_send_reply("404", "Not Found");

                 exit;

        };

        append_hf("P-hint: usrloc applied\r\n");

};

route(1);

}

 

route[1] {

        if (!t_relay()) {

                sl_reply_error();

        };

        exit;

       }

 

 

BUT when attempting a call am getting the message 'Call failed: Not found .'

 

 

Any help to achieve the successful call using above scenario is highly
appreciated.

 

Thanks in advance.

F.Chahrour

 

 

 

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From evilzluk at gmail.com  Wed Aug  8 18:07:24 2012
From: evilzluk at gmail.com (Konstantin M.)
Date: Wed, 8 Aug 2012 19:07:24 +0300
Subject: [SR-Users] Kernel Module Issue
In-Reply-To: <50224521.1030209@loopmethods.com>
References: <50224521.1030209@loopmethods.com>
Message-ID: <CAEYtTze5QSV7gcG0HXQfrhAtCuU=y6G4eUBVMR3ruKPNw+uP8A@mail.gmail.com>

echo "blacklist nf_conntrack_sip" >> /etc/modprobe.d/blacklist.conf


2012/8/8 Vijay Thakur <vijay.thakur at loopmethods.com>

>  Thanks for the hint. What is the file name in /etc/modprob.d/ where i am supposed to enter this module name to disable it.
>
> Thanks
>
> Vijay Tha
>
> Message: 1
> Date: Mon, 06 Aug 2012 09:01:03 -0400
> From: Richard Fuchs <rfuchs at sipwise.com> <rfuchs at sipwise.com>
> Subject: Re: [SR-Users] Kernel Droping SIP packet
> To: sr-users at lists.sip-router.org
> Message-ID: <501FC00F.6000604 at sipwise.com> <501FC00F.6000604 at sipwise.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On 08/06/12 06:08, Vijay Thakur wrote:
>
>  Hi all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine.
> When i make a call from Soft phone (X-Lite) to iphone, all is working
> fine. But in other case call from iphone to Softphone is not working,
> even not ringing. During checking the logs i am getting the error:
>
> Aug  3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
>
>  This is coming from nf_conntrack_sip, which is a netfilter connection
> tracking kernel module for SIP. I've never used it, but judging from
> what Google brings up, it seems to be very buggy. You should be able to
> just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work
> and/or if you want to keep it from auto-loading, you can blacklist it in*/etc/modprobe.d/* and then reboot.
>
> HTH
>
> copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited.
> -----------------------------------------------------------------------------------------------------------------
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From jmr.richardson at gmail.com  Wed Aug  8 19:13:57 2012
From: jmr.richardson at gmail.com (JR Richardson)
Date: Wed, 8 Aug 2012 12:13:57 -0500
Subject: [SR-Users] Porblem Starting kamailio on Debian Squeeze
Message-ID: <CA+U74VNAA9u44iRsA3445MV=oMd4=QxNd0mUPL7goL1uj=cxWw@mail.gmail.com>

Hi All,

I'm running redundant kamailio 3.0.4 servers in production, have been
for a long time with great success. They were installed on debian
Lenny.  One of my servers crashed.  I can't seem to do a debian lennyy
install because that version is archived now.  I'm trying to install
on debian squeeze but still using kamailio 3.0.4 for compatibility
reasons.  The install went pretty normal except during boot up,
kamailio starts before mysql and networking so it starts then exists.

Once the server is fully booted, kamailio starts with the init script
without error.

I have adjusted the init script to check for networking and mysql to
start first and performed an 'update-rc.d kamailio defaults' but still
no luck.

Any help will be appreciated.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses


From abalashov at evaristesys.com  Wed Aug  8 19:17:00 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 08 Aug 2012 13:17:00 -0400
Subject: [SR-Users] Porblem Starting kamailio on Debian Squeeze
In-Reply-To: <CA+U74VNAA9u44iRsA3445MV=oMd4=QxNd0mUPL7goL1uj=cxWw@mail.gmail.com>
References: <CA+U74VNAA9u44iRsA3445MV=oMd4=QxNd0mUPL7goL1uj=cxWw@mail.gmail.com>
Message-ID: <50229F0C.1030708@evaristesys.com>

When you restart the service, does it say "using dependency-based boot 
sequencing", or just restart?  In other words, does 'squeeze' use 
dependency-based boot sequencing?  I don't remember.

If it does, read this:

http://wiki.debian.org/LSBInitScripts/DependencyBasedBoot

On 08/08/2012 01:13 PM, JR Richardson wrote:

> Hi All,
>
> I'm running redundant kamailio 3.0.4 servers in production, have been
> for a long time with great success. They were installed on debian
> Lenny.  One of my servers crashed.  I can't seem to do a debian lennyy
> install because that version is archived now.  I'm trying to install
> on debian squeeze but still using kamailio 3.0.4 for compatibility
> reasons.  The install went pretty normal except during boot up,
> kamailio starts before mysql and networking so it starts then exists.
>
> Once the server is fully booted, kamailio starts with the init script
> without error.
>
> I have adjusted the init script to check for networking and mysql to
> start first and performed an 'update-rc.d kamailio defaults' but still
> no luck.
>
> Any help will be appreciated.
>
> Thanks.
>
> JR
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From abalashov at evaristesys.com  Wed Aug  8 19:18:58 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 08 Aug 2012 13:18:58 -0400
Subject: [SR-Users] Porblem Starting kamailio on Debian Squeeze
In-Reply-To: <CA+U74VNAA9u44iRsA3445MV=oMd4=QxNd0mUPL7goL1uj=cxWw@mail.gmail.com>
References: <CA+U74VNAA9u44iRsA3445MV=oMd4=QxNd0mUPL7goL1uj=cxWw@mail.gmail.com>
Message-ID: <50229F82.3000608@evaristesys.com>

Or, perhaps more to the point:

http://wiki.debian.org/LSBInitScripts/

Another option, if you're feeling lazy, don't care about LSB crap, and 
just want to solve your problem right now, is to just start Kamailio 
manually in /etc/init.d/bootmisc.sh.  That gets fired up after all the 
LSB init scripts run.

On 08/08/2012 01:13 PM, JR Richardson wrote:

> Hi All,
>
> I'm running redundant kamailio 3.0.4 servers in production, have been
> for a long time with great success. They were installed on debian
> Lenny.  One of my servers crashed.  I can't seem to do a debian lennyy
> install because that version is archived now.  I'm trying to install
> on debian squeeze but still using kamailio 3.0.4 for compatibility
> reasons.  The install went pretty normal except during boot up,
> kamailio starts before mysql and networking so it starts then exists.
>
> Once the server is fully booted, kamailio starts with the init script
> without error.
>
> I have adjusted the init script to check for networking and mysql to
> start first and performed an 'update-rc.d kamailio defaults' but still
> no luck.
>
> Any help will be appreciated.
>
> Thanks.
>
> JR
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From fchahrour at vanrise.com  Wed Aug  8 19:22:40 2012
From: fchahrour at vanrise.com (Fatima Chahrour~Vanrise Support)
Date: Wed, 8 Aug 2012 20:22:40 +0300
Subject: [SR-Users] Kamailio LCR Module test scenario
In-Reply-To: <01bd01cd757b$0543d090$0fcb71b0$@com>
References: <01bd01cd757b$0543d090$0fcb71b0$@com>
Message-ID: <01e301cd758a$6b17f040$4147d0c0$@com>

Attachment here!

 

From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fatima
Chahrour~Vanrise Support
Sent: Wednesday, August 08, 2012 6:32 PM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: [SR-Users] Kamailio LCR Module test scenario

 

Dears,

 

Am trying to apply Kamailio LCR feature lab test, after am finally able to
run Kamailio with no errors, am not able to reach my target clarified in
this following scenario:

Call 961312345 - route the call to 192.x.x.15 in first place and if failed
route the call to 192.x.x.10. I applied needed configuration in lcr tables
(attached image)  and the routing script is:

 

route{

       if (!mf_process_maxfwd_header("10")) {

               sl_send_reply("483","Too Many Hops");

               exit;

       };

       if (msg:len >= 2048 ) {

               sl_send_reply("513", "Message too big");

               exit;

       };

 

if (!method=="REGISTER")

        record_route();

 

if (loose_route()) {

        append_hf("P-hint: rr-enforced\r\n");

        route(1);

};

 

 

if (!uri==myself) {

                append_hf("P-hint: outbound\r\n");

        route(1);

};

 

if (is_method("INVITE"))

       {

              if (!load_gws(1)) {

              sl_send_reply("503", "Unable to load gateways");

              exit;

              }

}

 

if (uri==myself) {

        if (method=="REGISTER") { 

                 save("location");

                 exit;

        };

        lookup("aliases");

        if (!uri==myself) {

                 append_hf("P-hint: outbound alias\r\n");

                 route(1);

                        };

 

        # native SIP destinations are handled using ourUSRLOC DB

        if (!lookup("location")) {

                 sl_send_reply("404", "Not Found");

                 exit;

        };

        append_hf("P-hint: usrloc applied\r\n");

};

route(1);

}

 

route[1] {

        if (!t_relay()) {

                sl_reply_error();

        };

        exit;

       }

 

 

BUT when attempting a call am getting the message 'Call failed: Not found .'

 

 

Any help to achieve the successful call using above scenario is highly
appreciated.

 

Thanks in advance.

F.Chahrour

 

 

 

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From jmr.richardson at gmail.com  Wed Aug  8 19:34:30 2012
From: jmr.richardson at gmail.com (JR Richardson)
Date: Wed, 8 Aug 2012 12:34:30 -0500
Subject: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
Message-ID: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>

On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
> I'm running redundant kamailio 3.0.4 servers in production, have been
> for a long time with great success. They were installed on debian
> Lenny.  One of my servers crashed.  I can't seem to do a debian lennyy
> install because that version is archived now.  I'm trying to install
> on debian squeeze but still using kamailio 3.0.4 for compatibility
> reasons.  The install went pretty normal except during boot up,
> kamailio starts before mysql and networking so it starts then exists.
>
> Once the server is fully booted, kamailio starts with the init script
> without error.
>
> I have adjusted the init script to check for networking and mysql to
> start first and performed an 'update-rc.d kamailio defaults' but still
> no luck.
>
> Any help will be appreciated.
>
> Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses

Yep, this is an
http://wiki.debian.org/LSBInitScripts/DependencyBasedBoot issue.
Thanks Alex.

The old init scrip that ships with Kamailio 3.0.4 that I am using is
not 100% compatible so I pulled a hack out of the hat and inserted
'sleep 20' before of 'check_fork ()' in the kamailio init script.

Kamailio now waits long enough for all services to start before
launching, simple but effective.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses


From abalashov at evaristesys.com  Wed Aug  8 19:36:36 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 08 Aug 2012 13:36:36 -0400
Subject: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
In-Reply-To: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>
References: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>
Message-ID: <5022A3A4.4090505@evaristesys.com>

On 08/08/2012 01:34 PM, JR Richardson wrote:

> The old init scrip that ships with Kamailio 3.0.4 that I am using is
> not 100% compatible so I pulled a hack out of the hat and inserted
> 'sleep 20' before of 'check_fork ()' in the kamailio init script.
>
> Kamailio now waits long enough for all services to start before
> launching, simple but effective.

Ah, my preferred solution.  :-)

-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From pavel.klochan at gmail.com  Wed Aug  8 19:49:20 2012
From: pavel.klochan at gmail.com (Pavel Klochan)
Date: Wed, 8 Aug 2012 21:49:20 +0400
Subject: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
In-Reply-To: <5022A3A4.4090505@evaristesys.com>
References: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>
	<5022A3A4.4090505@evaristesys.com>
Message-ID: <CAG5bokO_yHOH4roVrfejdzhQWfLH9K23VLyE1+-kZEO8fDkQgg@mail.gmail.com>

Hi.
Another solution is to change priority in /etc/rc2.d/
I think it's more correct, because if you change kamailio ini script -
after update you will need to change /etc/init.d/kamailio script manually

On Wed, Aug 8, 2012 at 9:36 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> On 08/08/2012 01:34 PM, JR Richardson wrote:
>
>  The old init scrip that ships with Kamailio 3.0.4 that I am using is
>> not 100% compatible so I pulled a hack out of the hat and inserted
>> 'sleep 20' before of 'check_fork ()' in the kamailio init script.
>>
>> Kamailio now waits long enough for all services to start before
>> launching, simple but effective.
>>
>
> Ah, my preferred solution.  :-)
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>
> ______________________________**_________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
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From abalashov at evaristesys.com  Wed Aug  8 19:50:57 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 08 Aug 2012 13:50:57 -0400
Subject: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
In-Reply-To: <CAG5bokO_yHOH4roVrfejdzhQWfLH9K23VLyE1+-kZEO8fDkQgg@mail.gmail.com>
References: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>
	<5022A3A4.4090505@evaristesys.com>
	<CAG5bokO_yHOH4roVrfejdzhQWfLH9K23VLyE1+-kZEO8fDkQgg@mail.gmail.com>
Message-ID: <5022A701.2090301@evaristesys.com>

Not in dependency-based boot sequencing.  :-)

On 08/08/2012 01:49 PM, Pavel Klochan wrote:

> Hi.
> Another solution is to change priority in /etc/rc2.d/
> I think it's more correct, because if you change kamailio ini script -
> after update you will need to change /etc/init.d/kamailio script manually
>
> On Wed, Aug 8, 2012 at 9:36 PM, Alex Balashov <abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>> wrote:
>
>     On 08/08/2012 01:34 PM, JR Richardson wrote:
>
>         The old init scrip that ships with Kamailio 3.0.4 that I am using is
>         not 100% compatible so I pulled a hack out of the hat and inserted
>         'sleep 20' before of 'check_fork ()' in the kamailio init script.
>
>         Kamailio now waits long enough for all services to start before
>         launching, simple but effective.
>
>
>     Ah, my preferred solution.  :-)
>
>     --
>     Alex Balashov - Principal
>     Evariste Systems LLC
>     235 E Ponce de Leon Ave
>     Suite 106
>     Decatur, GA 30030
>     Tel: +1-678-954-0670
>     Fax: +1-404-961-1892
>     Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
>
>     _________________________________________________
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>     http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users
>     <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From jmr.richardson at gmail.com  Wed Aug  8 19:55:20 2012
From: jmr.richardson at gmail.com (JR Richardson)
Date: Wed, 8 Aug 2012 12:55:20 -0500
Subject: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
In-Reply-To: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>
References: <CA+U74VOL8sSf_ZqYR0P-qNXUYY5zZ_92Kd99SkGmOkCF3uKqmw@mail.gmail.com>
Message-ID: <CA+U74VOyDp_730AhzXTaz_=-Fuo69nO4MrPy=3dMN-yCe91rpA@mail.gmail.com>

On Wed, Aug 8, 2012 at 12:34 PM, JR Richardson <jmr.richardson at gmail.com> wrote:
> On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson <jmr.richardson at gmail.com> wrote:
>> Hi All,
>>
>> I'm running redundant kamailio 3.0.4 servers in production, have been
>> for a long time with great success. They were installed on debian
>> Lenny.  One of my servers crashed.  I can't seem to do a debian lennyy
>> install because that version is archived now.  I'm trying to install
>> on debian squeeze but still using kamailio 3.0.4 for compatibility
>> reasons.  The install went pretty normal except during boot up,
>> kamailio starts before mysql and networking so it starts then exists.
>>
>> Once the server is fully booted, kamailio starts with the init script
>> without error.
>>
>> I have adjusted the init script to check for networking and mysql to
>> start first and performed an 'update-rc.d kamailio defaults' but still
>> no luck.
>>
>> Any help will be appreciated.
>>
>> Thanks.
>>
>> JR
>> --
>> JR Richardson
>> Engineering for the Masses
>
> Yep, this is an
> http://wiki.debian.org/LSBInitScripts/DependencyBasedBoot issue.
> Thanks Alex.
>
> The old init scrip that ships with Kamailio 3.0.4 that I am using is
> not 100% compatible so I pulled a hack out of the hat and inserted
> 'sleep 20' before of 'check_fork ()' in the kamailio init script.
>
> Kamailio now waits long enough for all services to start before
> launching, simple but effective.
>

After digging in a bit more, reading the directions and such.  When
you copy over the init script and run 'update-rd.d kamailio defaults',
this doesn't actually do anything in Debian Squeeze.  The new LSB init
scripts use the 'insserv' command to set init priorities.  Proper
steps are such:
Copy over squeeze init script to /etc/ini.d/kamailio
Edit /etc/init.d/kamailio
Add 'mysql' to 'Required-Start:' line
----snip-----
### BEGIN INIT INFO
# Provides:          kamailio
# Required-Start:    $syslog $network $local_fs $time mysql
# Required-Stop:     $syslog $network $local_fs
# Default-Start:     2 3 4 5
# Default-Stop:      0 1 6
# Short-Description: Start the Kamailio SIP proxy server
# Description:       Start the Kamailio SIP proxy server
### END INIT INFO
----end snip-------
Then run 'insserv kamailio'
Check /etc/rc2.d/ and you will see kamailio set to start after mysql.

That did it for me, I took out the 'sleep 20' from my previous post.

Hope this helps

JR
-- 
JR Richardson
Engineering for the Masses


From keeling at akan-tech.com  Thu Aug  9 05:42:54 2012
From: keeling at akan-tech.com (Nathaniel L Keeling)
Date: Wed, 08 Aug 2012 22:42:54 -0500
Subject: [SR-Users] postgres in kamailio  3.3.0-9.1
In-Reply-To: <501B9105.3050508@gmail.com>
References: <50194C47.1030602@telegroup.com.ua> <50198ACE.7030909@gmail.com>
	<5019CADF.4070703@akan.net> <501A27D7.90107@gmail.com>
	<501B48C1.5020005@akan-tech.com> <501B9105.3050508@gmail.com>
Message-ID: <502331BE.2000304@akan-tech.com>

I am running Centos v5.5 and both openssl versions 0.9 and 1.0 are 
installed. I tried to uninstall version 0.9 but there were over 400 
packages that were dependent upon that version.

Thanks

Nathaniel

On 8/3/2012 3:51 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 8/3/12 5:42 AM, Nathaniel L Keeling wrote:
>> Yes but it was for an older version that was uninstalled. I changed 
>> the path to pull in the current version and still received the same 
>> error. I checked the version for openssl and the same thing was 
>> happening. I changed the path again and recompiled kamailio and still 
>> no change. I noticed when compiling kamailio, the tls.so module was 
>> complaining that the version of openssl was less then 1.0. The 
>> version that I installed is 1.0.1c.
> openssl libs can be installed with many versions at the same time, as 
> the package names differ -- so 0.9x and 1.x can exist installed at the 
> same time. Double check you don't have both installed, as some 
> applications out there require 0.9, others 1.0, they could get 
> installed due to dependencies.
>
> It may happen that opensuse build service used a specific openssl 
> version when building the rpms -- the rpm spec requires generic 
> packages openssl and openssl-devel for building.
>
> What is the version of your OS?
>
> Cheers,
> Daniel
>



From govoiper at gmail.com  Thu Aug  9 12:55:38 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 9 Aug 2012 15:55:38 +0500
Subject: [SR-Users] Fwd: Kamailio Presence with XCAP not working accordingly
In-Reply-To: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
Message-ID: <CAJUJwtik8By_7SJaCz-asW+_tKhbCrcjpJsJh05bKBKtDauUrA@mail.gmail.com>

Hi,

I've followed the tutorial on kab.asipto.com for presence using built-in
xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple

I'm using Kamailio version 3.3.1 and did minor changes in modparams and
rtpproxy function calls and the kamailio accepted the configurations file
posted on the page and started.

But the problem is that I don't get the presence status of the contacts
still. Please suggest what to look for and how to troubleshoot this.

I get this error on Jitsi -

ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can
not be read"

On other soft phones no error appears but no presence is exchanged. Only IM
messages get through if the contact is online.


Regards,
Sammy
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From govoiper at gmail.com  Thu Aug  9 12:30:03 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 9 Aug 2012 15:30:03 +0500
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
Message-ID: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>

Hi,

I've followed the tutorial on kab.asipto.com for presence using built-in
xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple

I'm using Kamailio version 3.3.1 and did minor changes in modparams and
rtpproxy function calls and the kamailio accepted the configurations file
posted on the page and started.

But the problem is that I don't get the presence status of the contacts
still. Please suggest what to look for and how to troubleshoot this.

I get this error on Jitsi - image attached.

ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can
not be read"

Regards,
Sammy
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From peter.dunkley at crocodile-rcs.com  Fri Aug 10 10:32:27 2012
From: peter.dunkley at crocodile-rcs.com (Peter Dunkley)
Date: Fri, 10 Aug 2012 09:32:27 +0100
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
Message-ID: <1344587547.1881.5.camel@pd-notebook-linux.croc.internal>

Hi,

The xcap-caps document is hard-coded in kamailio.cfg and provides a list
of the document types the XCAP server supports.  This part of the
configuration from the tutorial builds and returns the xcap-caps
document:

        		if($xcapuri(u=>auid)=="xcap-caps")
        		{
        			$var(xbody) =
        "<?xml version='1.0' encoding='UTF-8'?>
        <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
          <auids>
            <auid>rls-services</auid>
            <auid>pidf-manipulation</auid>
            <auid>xcap-caps</auid>
            <auid>resource-lists</auid>
            <auid>pres-rules</auid>
            <auid>org.openmobilealliance.pres-rules</auid>
          </auids>
          <extensions>
          </extensions>
          <namespaces>
            <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
            <namespace>urn:ietf:params:xml:ns:pidf</namespace>
            <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
            <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
            <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
          </namespaces>
        </xcap-caps>";
        			xhttp_reply("200", "ok", "application/xcap-caps+xml",
        					"$var(xbody)");
        			exit;
        		}

This tutorial has always worked for me in the past (although it has been
well over a year since I last used it), so it looks like either a
problem with Jitsi (which is unlikely) or some misconfiguration on the
Kamailio side.

A tcpdump of the traffic between Jitsi and Kamailio would help with
working out which side has the problem.

Regards,

Peter

On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:

> Hi,
> 
> 
> 
> I've followed the tutorial on kab.asipto.com for presence using
> built-in xcap
> server. http://kb.asipto.com/kamailio:presence:k31-made-simple
> 
> 
> I'm using Kamailio version 3.3.1 and did minor changes in modparams
> and rtpproxy function calls and the kamailio accepted the
> configurations file posted on the page and started.
> 
> 
> But the problem is that I don't get the presence status of the
> contacts still. Please suggest what to look for and how to
> troubleshoot this.
> 
> 
> I get this error on Jitsi - image attached.
> 
> 
> ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource
> can not be read"
> 
> 
> Regards,
> Sammy
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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From govoiper at gmail.com  Fri Aug 10 10:41:24 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 13:41:24 +0500
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
Message-ID: <CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>

Thanks ,

Yes this tutorial worked for me as well some 5/7 months ago with kamailio
3.1 but I'm on 3.3 now and tried the same configurations file. Jitsi on the
other hand is the only phone I found which shows me this error. I'm trying
with Ekiga - eyebeam or xlite didn't seem happy with the presence icons as
well..But common thing was that I could send the Chat messages successfully.

I'm continuously looking at the tcpdumps and I'll share those here in a
while.

Thanks for replying and taking interest.

Regard,
Sammy


On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley <
peter.dunkley at crocodile-rcs.com> wrote:

> **
> Hi,
>
> The xcap-caps document is hard-coded in kamailio.cfg and provides a list
> of the document types the XCAP server supports.  This part of the
> configuration from the tutorial builds and returns the xcap-caps document:
>
> 		if($xcapuri(u=>auid)=="xcap-caps")
> 		{
> 			$var(xbody) =
> "<?xml version='1.0' encoding='UTF-8'?>
> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>   <auids>
>     <auid>rls-services</auid>
>     <auid>pidf-manipulation</auid>
>     <auid>xcap-caps</auid>
>     <auid>resource-lists</auid>
>     <auid>pres-rules</auid>
>     <auid>org.openmobilealliance.pres-rules</auid>
>   </auids>
>   <extensions>
>   </extensions>
>   <namespaces>
>     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>   </namespaces>
> </xcap-caps>";
> 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
> 					"$var(xbody)");
> 			exit;
> 		}
>
>  This tutorial has always worked for me in the past (although it has been
> well over a year since I last used it), so it looks like either a problem
> with Jitsi (which is unlikely) or some misconfiguration on the Kamailio
> side.
>
> A tcpdump of the traffic between Jitsi and Kamailio would help with
> working out which side has the problem.
>
> Regards,
>
> Peter
>
>
> On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>
> Hi,
>
>
>  I've followed the tutorial on kab.asipto.com for presence using built-in
> xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
>
>
>
>  I'm using Kamailio version 3.3.1 and did minor changes in modparams and
> rtpproxy function calls and the kamailio accepted the configurations file
> posted on the page and started.
>
>
>
>  But the problem is that I don't get the presence status of the contacts
> still. Please suggest what to look for and how to troubleshoot this.
>
>
>
>  I get this error on Jitsi - image attached.
>
>
>
>  ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource
> can not be read"
>
>
>
>  Regards,
>
>  Sammy
>
>  _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From peter.dunkley at crocodile-rcs.com  Fri Aug 10 10:53:34 2012
From: peter.dunkley at crocodile-rcs.com (Peter Dunkley)
Date: Fri, 10 Aug 2012 09:53:34 +0100
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
Message-ID: <1344588814.1881.10.camel@pd-notebook-linux.croc.internal>

Hi,

XCAP uses HTTP (albeit still carried on port 5060 in this tutorial),
messaging and so on uses SIP.  These are different protocols and handled
in kamailio.cfg differently.

Last time I checked Xlite didn't support XCAP at all - so I don't think
that's going to work at all.  I can't see XCAP/XDMS listed as an Ekiga
feature either.

I am using presence and XCAP on Kamailio 3.3 (my own configuration, not
the one from the tutorial) and it works fine.

Regards,

Peter

On Fri, 2012-08-10 at 13:41 +0500, SamyGo wrote:

> Thanks ,
> 
> 
> 
> Yes this tutorial worked for me as well some 5/7 months ago with
> kamailio 3.1 but I'm on 3.3 now and tried the same configurations
> file. Jitsi on the other hand is the only phone I found which shows me
> this error. I'm trying with Ekiga - eyebeam or xlite didn't seem happy
> with the presence icons as well..But common thing was that I could
> send the Chat messages successfully.
> 
> 
> I'm continuously looking at the tcpdumps and I'll share those here in
> a while. 
> 
> 
> Thanks for replying and taking interest.
> 
> 
> Regard,
> Sammy
> 
> 
> 
> On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley
> <peter.dunkley at crocodile-rcs.com> wrote:
> 
>         Hi,
>         
>         The xcap-caps document is hard-coded in kamailio.cfg and
>         provides a list of the document types the XCAP server
>         supports.  This part of the configuration from the tutorial
>         builds and returns the xcap-caps document: 
>         
>                 		if($xcapuri(u=>auid)=="xcap-caps")
>                 		{
>                 			$var(xbody) =
>                 "<?xml version='1.0' encoding='UTF-8'?>
>                 <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>                   <auids>
>                     <auid>rls-services</auid>
>                     <auid>pidf-manipulation</auid>
>                     <auid>xcap-caps</auid>
>                     <auid>resource-lists</auid>
>                     <auid>pres-rules</auid>
>                     <auid>org.openmobilealliance.pres-rules</auid>
>                   </auids>
>                   <extensions>
>                   </extensions>
>                   <namespaces>
>                     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>                     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>                     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>                     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>                     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>                   </namespaces>
>                 </xcap-caps>";
>                 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
>                 					"$var(xbody)");
>                 			exit;
>                 		}
>         
>         This tutorial has always worked for me in the past (although
>         it has been well over a year since I last used it), so it
>         looks like either a problem with Jitsi (which is unlikely) or
>         some misconfiguration on the Kamailio side.
>         
>         A tcpdump of the traffic between Jitsi and Kamailio would help
>         with working out which side has the problem.
>         
>         Regards,
>         
>         Peter
>         
>         
>         
>         On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>         
>         > Hi,
>         > 
>         > I've followed the tutorial on kab.asipto.com for presence
>         > using built-in xcap
>         > server. http://kb.asipto.com/kamailio:presence:k31-made-simple
>         > 
>         > 
>         > I'm using Kamailio version 3.3.1 and did minor changes in
>         > modparams and rtpproxy function calls and the kamailio
>         > accepted the configurations file posted on the page and
>         > started.
>         > 
>         > 
>         > But the problem is that I don't get the presence status of
>         > the contacts still. Please suggest what to look for and how
>         > to troubleshoot this.
>         > 
>         > 
>         > I get this error on Jitsi - image attached.
>         > 
>         > 
>         > ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index
>         > resource can not be read"
>         > 
>         > 
>         > Regards,
>         > Sammy
>         > 
>         > _______________________________________________
>         > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>         > sr-users at lists.sip-router.org
>         > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         
>         
>         -- 
>         Peter Dunkley
>         Technical Director
>         Crocodile RCS Ltd
>         
>         
>         _______________________________________________
>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>         mailing list
>         sr-users at lists.sip-router.org
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         
> 
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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From govoiper at gmail.com  Fri Aug 10 11:16:21 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 14:16:21 +0500
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <1344588814.1881.10.camel@pd-notebook-linux.croc.internal>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
	<1344588814.1881.10.camel@pd-notebook-linux.croc.internal>
Message-ID: <CAJUJwtgUA-BpFfSfjMjdm-fcVcH97sz3x-_03sev-LW9vTLzJA@mail.gmail.com>

Yes, Ekiga isn't good with it either. Can you just point me to some
documentation and help me on that. I am sure I can get the presence
working, but do you know hwo can I manage an offline chat thing in this !
do I need a separate dedicated xcap server for this feature ?


On Fri, Aug 10, 2012 at 1:53 PM, Peter Dunkley <
peter.dunkley at crocodile-rcs.com> wrote:

> **
> Hi,
>
> XCAP uses HTTP (albeit still carried on port 5060 in this tutorial),
> messaging and so on uses SIP.  These are different protocols and handled in
> kamailio.cfg differently.
>
> Last time I checked Xlite didn't support XCAP at all - so I don't think
> that's going to work at all.  I can't see XCAP/XDMS listed as an Ekiga
> feature either.
>
> I am using presence and XCAP on Kamailio 3.3 (my own configuration, not
> the one from the tutorial) and it works fine.
>
> Regards,
>
> Peter
>
>
> On Fri, 2012-08-10 at 13:41 +0500, SamyGo wrote:
>
> Thanks ,
>
>
>
>  Yes this tutorial worked for me as well some 5/7 months ago with
> kamailio 3.1 but I'm on 3.3 now and tried the same configurations file.
> Jitsi on the other hand is the only phone I found which shows me this
> error. I'm trying with Ekiga - eyebeam or xlite didn't seem happy with the
> presence icons as well..But common thing was that I could send the Chat
> messages successfully.
>
>
>
>  I'm continuously looking at the tcpdumps and I'll share those here in a
> while.
>
>
>
>  Thanks for replying and taking interest.
>
>
>
>  Regard,
>
>  Sammy
>
>
>
>  On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley <
> peter.dunkley at crocodile-rcs.com> wrote:
>
>  Hi,
>
> The xcap-caps document is hard-coded in kamailio.cfg and provides a list
> of the document types the XCAP server supports.  This part of the
> configuration from the tutorial builds and returns the xcap-caps document:
>
> 		if($xcapuri(u=>auid)=="xcap-caps")
> 		{
> 			$var(xbody) =
> "<?xml version='1.0' encoding='UTF-8'?>
> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>   <auids>
>     <auid>rls-services</auid>
>     <auid>pidf-manipulation</auid>
>     <auid>xcap-caps</auid>
>     <auid>resource-lists</auid>
>     <auid>pres-rules</auid>
>     <auid>org.openmobilealliance.pres-rules</auid>
>   </auids>
>   <extensions>
>   </extensions>
>   <namespaces>
>     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>   </namespaces>
> </xcap-caps>";
> 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
> 					"$var(xbody)");
> 			exit;
> 		}
>
>  This tutorial has always worked for me in the past (although it has been
> well over a year since I last used it), so it looks like either a problem
> with Jitsi (which is unlikely) or some misconfiguration on the Kamailio
> side.
>
> A tcpdump of the traffic between Jitsi and Kamailio would help with
> working out which side has the problem.
>
> Regards,
>
> Peter
>
>
>
> On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>
> Hi,
>
> I've followed the tutorial on kab.asipto.com for presence using built-in
> xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
>
>
> I'm using Kamailio version 3.3.1 and did minor changes in modparams and
> rtpproxy function calls and the kamailio accepted the configurations file
> posted on the page and started.
>
>
> But the problem is that I don't get the presence status of the contacts
> still. Please suggest what to look for and how to troubleshoot this.
>
>
> I get this error on Jitsi - image attached.
>
>
> ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can
> not be read"
>
>
> Regards,
> Sammy
>
>    _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>  _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From peter.dunkley at crocodile-rcs.com  Fri Aug 10 12:00:28 2012
From: peter.dunkley at crocodile-rcs.com (Peter Dunkley)
Date: Fri, 10 Aug 2012 11:00:28 +0100
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <CAJUJwtgUA-BpFfSfjMjdm-fcVcH97sz3x-_03sev-LW9vTLzJA@mail.gmail.com>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
	<1344588814.1881.10.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtgUA-BpFfSfjMjdm-fcVcH97sz3x-_03sev-LW9vTLzJA@mail.gmail.com>
Message-ID: <1344592828.1881.14.camel@pd-notebook-linux.croc.internal>

Hi,

The only documentation I know of is the stuff in the tutorial, the RFCs,
and the Kamailio module documentation.

Offline chat is a SIP feature, it has nothing to do with the XCAP
server.  The XCAP server manages XML documents (presence authorisation
rules, contact lists, avatars, etc) and is accessed using HTTP.  Instant
messages are SIP MESSAGE requests and are handled by the SIP routing
part of Kamailio.

If you want to do offline message handling you need the msilo Kamailio
module (which isn't part of the tutorial).

Regards,

Peter

On Fri, 2012-08-10 at 14:16 +0500, SamyGo wrote:

> Yes, Ekiga isn't good with it either. Can you just point me to some
> documentation and help me on that. I am sure I can get the presence
> working, but do you know hwo can I manage an offline chat thing in
> this ! do I need a separate dedicated xcap server for this feature ?
> 
> 
> 
> 
> On Fri, Aug 10, 2012 at 1:53 PM, Peter Dunkley
> <peter.dunkley at crocodile-rcs.com> wrote:
> 
>         Hi,
>         
>         XCAP uses HTTP (albeit still carried on port 5060 in this
>         tutorial), messaging and so on uses SIP.  These are different
>         protocols and handled in kamailio.cfg differently.
>         
>         Last time I checked Xlite didn't support XCAP at all - so I
>         don't think that's going to work at all.  I can't see
>         XCAP/XDMS listed as an Ekiga feature either.
>         
>         I am using presence and XCAP on Kamailio 3.3 (my own
>         configuration, not the one from the tutorial) and it works
>         fine.
>         
>         Regards,
>         
>         Peter
>         
>         
>         
>         On Fri, 2012-08-10 at 13:41 +0500, SamyGo wrote:
>         
>         > Thanks ,
>         > 
>         > 
>         > Yes this tutorial worked for me as well some 5/7 months ago
>         > with kamailio 3.1 but I'm on 3.3 now and tried the same
>         > configurations file. Jitsi on the other hand is the only
>         > phone I found which shows me this error. I'm trying with
>         > Ekiga - eyebeam or xlite didn't seem happy with the presence
>         > icons as well..But common thing was that I could send the
>         > Chat messages successfully.
>         > 
>         > 
>         > I'm continuously looking at the tcpdumps and I'll share
>         > those here in a while. 
>         > 
>         > 
>         > Thanks for replying and taking interest.
>         > 
>         > 
>         > Regard,
>         > Sammy
>         > 
>         > 
>         > On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley
>         > <peter.dunkley at crocodile-rcs.com> wrote:
>         > 
>         >         Hi,
>         >         
>         >         The xcap-caps document is hard-coded in kamailio.cfg
>         >         and provides a list of the document types the XCAP
>         >         server supports.  This part of the configuration
>         >         from the tutorial builds and returns the xcap-caps
>         >         document: 
>         >         
>         >                 		if($xcapuri(u=>auid)=="xcap-caps")
>         >                 		{
>         >                 			$var(xbody) =
>         >                 "<?xml version='1.0' encoding='UTF-8'?>
>         >                 <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>         >                   <auids>
>         >                     <auid>rls-services</auid>
>         >                     <auid>pidf-manipulation</auid>
>         >                     <auid>xcap-caps</auid>
>         >                     <auid>resource-lists</auid>
>         >                     <auid>pres-rules</auid>
>         >                     <auid>org.openmobilealliance.pres-rules</auid>
>         >                   </auids>
>         >                   <extensions>
>         >                   </extensions>
>         >                   <namespaces>
>         >                     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>         >                     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>         >                     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>         >                     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>         >                     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>         >                   </namespaces>
>         >                 </xcap-caps>";
>         >                 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
>         >                 					"$var(xbody)");
>         >                 			exit;
>         >                 		}
>         >         
>         >         This tutorial has always worked for me in the past
>         >         (although it has been well over a year since I last
>         >         used it), so it looks like either a problem with
>         >         Jitsi (which is unlikely) or some misconfiguration
>         >         on the Kamailio side.
>         >         
>         >         A tcpdump of the traffic between Jitsi and Kamailio
>         >         would help with working out which side has the
>         >         problem.
>         >         
>         >         Regards,
>         >         
>         >         Peter 
>         >         
>         >         
>         >         On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>         >         
>         >         > Hi,
>         >         > 
>         >         > I've followed the tutorial on kab.asipto.com for
>         >         > presence using built-in xcap
>         >         > server. http://kb.asipto.com/kamailio:presence:k31-made-simple
>         >         > 
>         >         > 
>         >         > I'm using Kamailio version 3.3.1 and did minor
>         >         > changes in modparams and rtpproxy function calls
>         >         > and the kamailio accepted the configurations file
>         >         > posted on the page and started.
>         >         > 
>         >         > 
>         >         > But the problem is that I don't get the presence
>         >         > status of the contacts still. Please suggest what
>         >         > to look for and how to troubleshoot this.
>         >         > 
>         >         > 
>         >         > I get this error on Jitsi - image attached.
>         >         > 
>         >         > 
>         >         > ERROR:
>         >         > "http://ip.of.server/xcap-root/xcap-caps/global/index resource can not be read"
>         >         > 
>         >         > 
>         >         > Regards,
>         >         > Sammy 
>         >         > 
>         >         > _______________________________________________
>         >         > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>         >         > sr-users at lists.sip-router.org
>         >         > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         >         
>         >         
>         >         -- 
>         >         Peter Dunkley
>         >         Technical Director
>         >         Crocodile RCS Ltd
>         >         
>         >         _______________________________________________
>         >         SIP Express Router (SER) and Kamailio (OpenSER) -
>         >         sr-users mailing list
>         >         sr-users at lists.sip-router.org
>         >         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         >         
>         > 
>         > 
>         > 
>         > 
>         > _______________________________________________
>         > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>         > sr-users at lists.sip-router.org
>         > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         
>         
>         -- 
>         Peter Dunkley
>         Technical Director
>         Crocodile RCS Ltd
>         
>         
>         _______________________________________________
>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>         mailing list
>         sr-users at lists.sip-router.org
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>         
> 
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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From govoiper at gmail.com  Fri Aug 10 12:04:46 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 15:04:46 +0500
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <1344592828.1881.14.camel@pd-notebook-linux.croc.internal>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
	<1344588814.1881.10.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtgUA-BpFfSfjMjdm-fcVcH97sz3x-_03sev-LW9vTLzJA@mail.gmail.com>
	<1344592828.1881.14.camel@pd-notebook-linux.croc.internal>
Message-ID: <CAJUJwtgST=jfYQkBjLUb_ScJPND+SkXRooFfs+gXE9POPFkyWQ@mail.gmail.com>

Thats good enough for me, I'll go through everything and see if I can make
it work.
Thanks for your help and time.


On Fri, Aug 10, 2012 at 3:00 PM, Peter Dunkley <
peter.dunkley at crocodile-rcs.com> wrote:

> **
> Hi,
>
> The only documentation I know of is the stuff in the tutorial, the RFCs,
> and the Kamailio module documentation.
>
> Offline chat is a SIP feature, it has nothing to do with the XCAP server.
> The XCAP server manages XML documents (presence authorisation rules,
> contact lists, avatars, etc) and is accessed using HTTP.  Instant messages
> are SIP MESSAGE requests and are handled by the SIP routing part of
> Kamailio.
>
> If you want to do offline message handling you need the msilo Kamailio
> module (which isn't part of the tutorial).
>
> Regards,
>
> Peter
>
>
> On Fri, 2012-08-10 at 14:16 +0500, SamyGo wrote:
>
> Yes, Ekiga isn't good with it either. Can you just point me to some
> documentation and help me on that. I am sure I can get the presence
> working, but do you know hwo can I manage an offline chat thing in this !
> do I need a separate dedicated xcap server for this feature ?
>
>
>
>  On Fri, Aug 10, 2012 at 1:53 PM, Peter Dunkley <
> peter.dunkley at crocodile-rcs.com> wrote:
>
>  Hi,
>
> XCAP uses HTTP (albeit still carried on port 5060 in this tutorial),
> messaging and so on uses SIP.  These are different protocols and handled in
> kamailio.cfg differently.
>
> Last time I checked Xlite didn't support XCAP at all - so I don't think
> that's going to work at all.  I can't see XCAP/XDMS listed as an Ekiga
> feature either.
>
> I am using presence and XCAP on Kamailio 3.3 (my own configuration, not
> the one from the tutorial) and it works fine.
>
> Regards,
>
> Peter
>
>
>
> On Fri, 2012-08-10 at 13:41 +0500, SamyGo wrote:
>
> Thanks ,
>
>
> Yes this tutorial worked for me as well some 5/7 months ago with kamailio
> 3.1 but I'm on 3.3 now and tried the same configurations file. Jitsi on the
> other hand is the only phone I found which shows me this error. I'm trying
> with Ekiga - eyebeam or xlite didn't seem happy with the presence icons as
> well..But common thing was that I could send the Chat messages successfully.
>
>
> I'm continuously looking at the tcpdumps and I'll share those here in a
> while.
>
>
> Thanks for replying and taking interest.
>
>
> Regard,
> Sammy
>
>
> On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley <
> peter.dunkley at crocodile-rcs.com> wrote:
>
> Hi,
>
> The xcap-caps document is hard-coded in kamailio.cfg and provides a list
> of the document types the XCAP server supports.  This part of the
> configuration from the tutorial builds and returns the xcap-caps document:
>
> 		if($xcapuri(u=>auid)=="xcap-caps")
> 		{
> 			$var(xbody) =
> "<?xml version='1.0' encoding='UTF-8'?>
> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>   <auids>
>     <auid>rls-services</auid>
>     <auid>pidf-manipulation</auid>
>     <auid>xcap-caps</auid>
>     <auid>resource-lists</auid>
>     <auid>pres-rules</auid>
>     <auid>org.openmobilealliance.pres-rules</auid>
>   </auids>
>   <extensions>
>   </extensions>
>   <namespaces>
>     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>   </namespaces>
> </xcap-caps>";
> 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
> 					"$var(xbody)");
> 			exit;
> 		}
>
>  This tutorial has always worked for me in the past (although it has been
> well over a year since I last used it), so it looks like either a problem
> with Jitsi (which is unlikely) or some misconfiguration on the Kamailio
> side.
>
> A tcpdump of the traffic between Jitsi and Kamailio would help with
> working out which side has the problem.
>
> Regards,
>
> Peter
>
>
> On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>
> Hi,
>
> I've followed the tutorial on kab.asipto.com for presence using built-in
> xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
>
>
> I'm using Kamailio version 3.3.1 and did minor changes in modparams and
> rtpproxy function calls and the kamailio accepted the configurations file
> posted on the page and started.
>
>
> But the problem is that I don't get the presence status of the contacts
> still. Please suggest what to look for and how to troubleshoot this.
>
>
> I get this error on Jitsi - image attached.
>
>
> ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can
> not be read"
>
>
> Regards,
> Sammy
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>  _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From govoiper at gmail.com  Fri Aug 10 12:41:50 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 15:41:50 +0500
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <CAJUJwtgST=jfYQkBjLUb_ScJPND+SkXRooFfs+gXE9POPFkyWQ@mail.gmail.com>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
	<1344588814.1881.10.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtgUA-BpFfSfjMjdm-fcVcH97sz3x-_03sev-LW9vTLzJA@mail.gmail.com>
	<1344592828.1881.14.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtgST=jfYQkBjLUb_ScJPND+SkXRooFfs+gXE9POPFkyWQ@mail.gmail.com>
Message-ID: <CAJUJwtir5EpX0_a7Q_rB_G=mO6Sk2B1cuV4Ee6tP85hQpuGyiA@mail.gmail.com>

Please see the SIP capture. As I was changing the online status from my
jitis and eyebeam phone I could see the publish requests handled by
kamailio but isn't it strange that I don'tfind any Notify generated and
relayed to the watchers ?


BR
Sammy


On Fri, Aug 10, 2012 at 3:04 PM, SamyGo <govoiper at gmail.com> wrote:

> Thats good enough for me, I'll go through everything and see if I can make
> it work.
> Thanks for your help and time.
>
>
> On Fri, Aug 10, 2012 at 3:00 PM, Peter Dunkley <
> peter.dunkley at crocodile-rcs.com> wrote:
>
>> **
>> Hi,
>>
>> The only documentation I know of is the stuff in the tutorial, the RFCs,
>> and the Kamailio module documentation.
>>
>> Offline chat is a SIP feature, it has nothing to do with the XCAP
>> server.  The XCAP server manages XML documents (presence authorisation
>> rules, contact lists, avatars, etc) and is accessed using HTTP.  Instant
>> messages are SIP MESSAGE requests and are handled by the SIP routing part
>> of Kamailio.
>>
>> If you want to do offline message handling you need the msilo Kamailio
>> module (which isn't part of the tutorial).
>>
>> Regards,
>>
>> Peter
>>
>>
>> On Fri, 2012-08-10 at 14:16 +0500, SamyGo wrote:
>>
>> Yes, Ekiga isn't good with it either. Can you just point me to some
>> documentation and help me on that. I am sure I can get the presence
>> working, but do you know hwo can I manage an offline chat thing in this !
>> do I need a separate dedicated xcap server for this feature ?
>>
>>
>>
>>  On Fri, Aug 10, 2012 at 1:53 PM, Peter Dunkley <
>> peter.dunkley at crocodile-rcs.com> wrote:
>>
>>  Hi,
>>
>> XCAP uses HTTP (albeit still carried on port 5060 in this tutorial),
>> messaging and so on uses SIP.  These are different protocols and handled in
>> kamailio.cfg differently.
>>
>> Last time I checked Xlite didn't support XCAP at all - so I don't think
>> that's going to work at all.  I can't see XCAP/XDMS listed as an Ekiga
>> feature either.
>>
>> I am using presence and XCAP on Kamailio 3.3 (my own configuration, not
>> the one from the tutorial) and it works fine.
>>
>> Regards,
>>
>> Peter
>>
>>
>>
>> On Fri, 2012-08-10 at 13:41 +0500, SamyGo wrote:
>>
>> Thanks ,
>>
>>
>> Yes this tutorial worked for me as well some 5/7 months ago with kamailio
>> 3.1 but I'm on 3.3 now and tried the same configurations file. Jitsi on the
>> other hand is the only phone I found which shows me this error. I'm trying
>> with Ekiga - eyebeam or xlite didn't seem happy with the presence icons as
>> well..But common thing was that I could send the Chat messages successfully.
>>
>>
>> I'm continuously looking at the tcpdumps and I'll share those here in a
>> while.
>>
>>
>> Thanks for replying and taking interest.
>>
>>
>> Regard,
>> Sammy
>>
>>
>> On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley <
>> peter.dunkley at crocodile-rcs.com> wrote:
>>
>> Hi,
>>
>> The xcap-caps document is hard-coded in kamailio.cfg and provides a list
>> of the document types the XCAP server supports.  This part of the
>> configuration from the tutorial builds and returns the xcap-caps document:
>>
>> 		if($xcapuri(u=>auid)=="xcap-caps")
>> 		{
>> 			$var(xbody) =
>> "<?xml version='1.0' encoding='UTF-8'?>
>> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>>   <auids>
>>     <auid>rls-services</auid>
>>     <auid>pidf-manipulation</auid>
>>     <auid>xcap-caps</auid>
>>     <auid>resource-lists</auid>
>>     <auid>pres-rules</auid>
>>     <auid>org.openmobilealliance.pres-rules</auid>
>>   </auids>
>>   <extensions>
>>   </extensions>
>>   <namespaces>
>>     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>>     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>>     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>>     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>>     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>>   </namespaces>
>> </xcap-caps>";
>> 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
>> 					"$var(xbody)");
>> 			exit;
>> 		}
>>
>>  This tutorial has always worked for me in the past (although it has been
>> well over a year since I last used it), so it looks like either a problem
>> with Jitsi (which is unlikely) or some misconfiguration on the Kamailio
>> side.
>>
>> A tcpdump of the traffic between Jitsi and Kamailio would help with
>> working out which side has the problem.
>>
>> Regards,
>>
>> Peter
>>
>>
>> On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>>
>> Hi,
>>
>> I've followed the tutorial on kab.asipto.com for presence using built-in
>> xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
>>
>>
>> I'm using Kamailio version 3.3.1 and did minor changes in modparams and
>> rtpproxy function calls and the kamailio accepted the configurations file
>> posted on the page and started.
>>
>>
>> But the problem is that I don't get the presence status of the contacts
>> still. Please suggest what to look for and how to troubleshoot this.
>>
>>
>> I get this error on Jitsi - image attached.
>>
>>
>> ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource
>> can not be read"
>>
>>
>> Regards,
>> Sammy
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>   --
>> Peter Dunkley
>> Technical Director
>> Crocodile RCS Ltd
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>   --
>> Peter Dunkley
>> Technical Director
>> Crocodile RCS Ltd
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>  _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>   --
>> Peter Dunkley
>> Technical Director
>> Crocodile RCS Ltd
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
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From ushacked at gmail.com  Sun Aug 12 13:50:55 2012
From: ushacked at gmail.com (Uri Shacked)
Date: Sun, 12 Aug 2012 14:50:55 +0300
Subject: [SR-Users] $avp reset to <null> with no reason?
Message-ID: <CAMMbDhQMMqM9GSGFjBGkH16N-wb6g=z0EfdYoy5AP59uYF-7Yg@mail.gmail.com>

Hi,****

** **

I encountered a problem. While doing some avp's settings I notice that one
avp that I use is changing to <null> without me doing anything.****

** **

Here is the part of the cfg file:****

????.****

????.****

Route[SET]{****

????..****

$avp(Sfeaturetype)=$avp(next_feature_type);****

????****

Route(DIDSRV);****

}****

** **

route[DIDSRV] {****

xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV type = $avp(Sfeaturetype), index =
$avp(SfeatureIndx) $C(xx)\n");****

if($var(srvcount)>3){****

xlog("L_NOTICE","$C(rg)Too many services loops $C(xx)\n");****

update_stat("nts_subs_srv_loop", "+1");****

$avp(TRMCS)="10";****

t_reply(403,"too many loops");****

exit;****

}****

$var(srvcount)=($var(srvcount)+1);****

xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV before switch to $avp(SfeatureType)
or $avp(next_feature_type) $C(xx)\n");****

switch ($avp(SfeatureType)){****

case "0" :****

???.****

???****

** **

On the route "SET" I put the value from the $avp(next_feature_type) in
$avp(Sfeaturetype).****

After that I call the route "DIDSRV" and use switch($avp(Sfeaturetype)).
Before the switch I xlog the value of the avp twice.****

As you can see in the log below, on the first xlog print, the value is 0
(as I expect). On the second print it resets to <null>****

WHY?****

** **

Log:****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:413]: qm_malloc(0x7fe6f9aff000, 72) returns address
0x7fe6fcc4d330 frag. 0x7fe6fcc4d300 (size=72) on 1 -th hit****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
#033[0;31;42m SCRIPT: before DIDSRV type = 0, index = 8888888889
#033[0;39;49m****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
#033[0;31;42m SCRIPT*: in DIDSRV type = 0*, index = 8888888889
#033<8888888889%C2%A0%20%23033>
[0;39;49m****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 848) called from <core>:
rvalue.c: rval_new_empty(236)****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 848) returns address
0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from
<core>: rvalue.c: rval_destroy(141)****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from
<core>: rvalue.c: rval_new_empty(236)****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
#033[0;31;42m SCRIPT*: in DIDSRV before switch to <null>* or 0 #033[0;39;49m
****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 928) called from <core>:
rvalue.c: rval_new_empty(236)****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 928) returns address
0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from
<core>: rvalue.c: rval_destroy(141)****

Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from
<core>: rvalue.c: rval_new_empty(236)
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From fchahrour at vanrise.com  Sun Aug 12 16:09:45 2012
From: fchahrour at vanrise.com (Fatima Chahrour~ Vanrise Technical Support)
Date: Sun, 12 Aug 2012 17:09:45 +0300
Subject: [SR-Users] Kamailio LCR Module test scenario
In-Reply-To: <01e301cd758a$6b17f040$4147d0c0$@com>
References: <01bd01cd757b$0543d090$0fcb71b0$@com>
	<01e301cd758a$6b17f040$4147d0c0$@com>
Message-ID: <029d01cd7894$22acb810$68062830$@vanrise.com>

The below problem was solved after modifying the related routing logic.

 

Thanks!

 

From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fatima
Chahrour~Vanrise Support
Sent: Wednesday, August 08, 2012 8:23 PM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Re: [SR-Users] Kamailio LCR Module test scenario

 

Attachment here!

 

From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fatima
Chahrour~Vanrise Support
Sent: Wednesday, August 08, 2012 6:32 PM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: [SR-Users] Kamailio LCR Module test scenario

 

Dears,

 

Am trying to apply Kamailio LCR feature lab test, after am finally able to
run Kamailio with no errors, am not able to reach my target clarified in
this following scenario:

Call 961312345 - route the call to 192.x.x.15 in first place and if failed
route the call to 192.x.x.10. I applied needed configuration in lcr tables
(attached image)  and the routing script is:

 

route{

       if (!mf_process_maxfwd_header("10")) {

               sl_send_reply("483","Too Many Hops");

               exit;

       };

       if (msg:len >= 2048 ) {

               sl_send_reply("513", "Message too big");

               exit;

       };

 

if (!method=="REGISTER")

        record_route();

 

if (loose_route()) {

        append_hf("P-hint: rr-enforced\r\n");

        route(1);

};

 

 

if (!uri==myself) {

                append_hf("P-hint: outbound\r\n");

        route(1);

};

 

if (is_method("INVITE"))

       {

              if (!load_gws(1)) {

              sl_send_reply("503", "Unable to load gateways");

              exit;

              }

}

 

if (uri==myself) {

        if (method=="REGISTER") { 

                 save("location");

                 exit;

        };

        lookup("aliases");

        if (!uri==myself) {

                 append_hf("P-hint: outbound alias\r\n");

                 route(1);

                        };

 

        # native SIP destinations are handled using ourUSRLOC DB

        if (!lookup("location")) {

                 sl_send_reply("404", "Not Found");

                 exit;

        };

        append_hf("P-hint: usrloc applied\r\n");

};

route(1);

}

 

route[1] {

        if (!t_relay()) {

                sl_reply_error();

        };

        exit;

       }

 

 

BUT when attempting a call am getting the message 'Call failed: Not found .'

 

 

Any help to achieve the successful call using above scenario is highly
appreciated.

 

Thanks in advance.

F.Chahrour

 

 

 

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From anton.jugatsu at gmail.com  Mon Aug 13 07:41:07 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Mon, 13 Aug 2012 09:41:07 +0400
Subject: [SR-Users] Best tool for crafting sip messages
Message-ID: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>

Hello, guys.

What is the best tool for crafting sip messages for testing purpose? For
example, simple REGISTER or INVITE message. I'm new to kamailio (not sip)
so it would be great for me to use debbuger module for step-by-step digging
into cfg.
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From jgallart at systemonenoc.com  Mon Aug 13 08:05:46 2012
From: jgallart at systemonenoc.com (Javi Gallart)
Date: Mon, 13 Aug 2012 08:05:46 +0200
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
Message-ID: <02516738-A2B8-41CF-AC9B-1114E8AF000A@systemonenoc.com>

Hello

you definitely wan to to try sipp (http://sipp.sourceforge.net/). I  
haven't used sip inspector but have heard good things about it (http://www.sipinspector.com/ 
).

Regards

Javi
On Aug 13, 2012, at 7:41 AM, Anton Kvashenkin wrote:

> Hello, guys.
>
> What is the best tool for crafting sip messages for testing purpose?  
> For example, simple REGISTER or INVITE message. I'm new to kamailio  
> (not sip) so it would be great for me to use debbuger module for  
> step-by-step digging into cfg.
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing  
> list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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From anton.jugatsu at gmail.com  Mon Aug 13 08:35:18 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Mon, 13 Aug 2012 10:35:18 +0400
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <02516738-A2B8-41CF-AC9B-1114E8AF000A@systemonenoc.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
	<02516738-A2B8-41CF-AC9B-1114E8AF000A@systemonenoc.com>
Message-ID: <CANBhyPBDaFqFGB332-K57A1VnhoSnFxVE_4ZZL8baa1LmwBuKg@mail.gmail.com>

Thanks for the sipinspector. I will try it.

2012/8/13 Javi Gallart <jgallart at systemonenoc.com>

> Hello
>
> you definitely wan to to try sipp (http://sipp.sourceforge.net/). I
> haven't used sip inspector but have heard good things about it (
> http://www.sipinspector.com/).
>
> Regards
>
> Javi
> On Aug 13, 2012, at 7:41 AM, Anton Kvashenkin wrote:
>
> Hello, guys.
>
> What is the best tool for crafting sip messages for testing purpose? For
> example, simple REGISTER or INVITE message. I'm new to kamailio (not sip)
> so it would be great for me to use debbuger module for step-by-step digging
> into cfg.
>  _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From brandon at cryy.com  Mon Aug 13 08:37:01 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Sun, 12 Aug 2012 23:37:01 -0700
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
Message-ID: <17400764-B488-4C3F-9D63-5C54BAAB08B6@cryy.com>

Sipsak

Sent from my iPhone

On Aug 12, 2012, at 10:41 PM, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:

> Hello, guys. 
> 
> What is the best tool for crafting sip messages for testing purpose? For example, simple REGISTER or INVITE message. I'm new to kamailio (not sip) so it would be great for me to use debbuger module for step-by-step digging into cfg.
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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From anton.jugatsu at gmail.com  Mon Aug 13 08:45:39 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Mon, 13 Aug 2012 10:45:39 +0400
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <17400764-B488-4C3F-9D63-5C54BAAB08B6@cryy.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
	<17400764-B488-4C3F-9D63-5C54BAAB08B6@cryy.com>
Message-ID: <CANBhyPCxuZVtr5O87q0UWh-fTcRp8=wK2+tq+X1eCXQCbAjBzg@mail.gmail.com>

sipsak is very handy tool but AFAIK it lacks of simulating call flow like
REGISTER, 401 etc. Only -f register.message.

2012/8/13 Brandon Armstead <brandon at cryy.com>

> Sipsak
>
> Sent from my iPhone
>
> On Aug 12, 2012, at 10:41 PM, Anton Kvashenkin <anton.jugatsu at gmail.com>
> wrote:
>
> Hello, guys.
>
> What is the best tool for crafting sip messages for testing purpose? For
> example, simple REGISTER or INVITE message. I'm new to kamailio (not sip)
> so it would be great for me to use debbuger module for step-by-step digging
> into cfg.
>
> _______________________________________________
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From mark at delfin.me  Mon Aug 13 07:45:31 2012
From: mark at delfin.me (Mark Anthony Delfin)
Date: Mon, 13 Aug 2012 15:45:31 +1000
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
Message-ID: <CADAvG3+UvGcWvPpe5zrrHTv2hA47V-h5Z-OLtuxAAhF6d9LADQ@mail.gmail.com>

Hi Anton,

Previously I used the following.

sipsak
http://sipsak.org/

or

sipp
http://sipp.sourceforge.net/

Regards,

Mark
On Mon, Aug 13, 2012 at 3:41 PM, Anton Kvashenkin
<anton.jugatsu at gmail.com>wrote:

> Hello, guys.
>
> What is the best tool for crafting sip messages for testing purpose? For
> example, simple REGISTER or INVITE message. I'm new to kamailio (not sip)
> so it would be great for me to use debbuger module for step-by-step digging
> into cfg.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From apogrebennyk at sipwise.com  Mon Aug 13 09:03:07 2012
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Mon, 13 Aug 2012 09:03:07 +0200
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <CADAvG3+UvGcWvPpe5zrrHTv2hA47V-h5Z-OLtuxAAhF6d9LADQ@mail.gmail.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
	<CADAvG3+UvGcWvPpe5zrrHTv2hA47V-h5Z-OLtuxAAhF6d9LADQ@mail.gmail.com>
Message-ID: <5028A6AB.3060302@sipwise.com>

Same here, also you can use protoshoot provided by kamailio, see
utils/protoshoot.

On 08/13/2012 07:45 AM, Mark Anthony Delfin wrote:
> Hi Anton,
> 
> Previously I used the following.
> 
> sipsak
> http://sipsak.org/
> 
> or
> 
> sipp
> http://sipp.sourceforge.net/
> 
> Regards,
> 
> Mark



From phillman25 at gmail.com  Mon Aug 13 09:52:04 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 13 Aug 2012 10:52:04 +0300
Subject: [SR-Users] Remove m=video from SDP header
Message-ID: <CAEprJzK3qJfU-_A-OPYjRq+iHnYjimTv-u3jrup7i0rDwxA4-g@mail.gmail.com>

Dear List

I am trying to remove specific lines from the following original SDP body:


Content-Type: application/sdp
Content-Length: 406

v=0
o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 11904 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 11602 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv




I need to remove the following lines from the above SDP body as my
International carrier does not support Video capabilities:

m=video 15042 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000



I used the following code in Kamailio config:

if(has_body("application/sdp") && search_body("m=video")){

                   subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #');

                   subst_body('#a=rtpmap:34 (.*)$# #');

                   subst_body('#a=rtpmap:99 (.*)$# #');

                   subst_body('#a=sendrecv(.*)$# #');

        }




Content-Type: application/sdp
Content-Length: 325
P-hint: outbound
v=0
o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 11904 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


As you can see the m=video body has been removed, however, calls are still
failing. Is there something i have missed?

I am using Kamailio v3.2.2


Thanking you in advance!

Phillip
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From ushacked at gmail.com  Mon Aug 13 11:16:42 2012
From: ushacked at gmail.com (Uri Shacked)
Date: Mon, 13 Aug 2012 12:16:42 +0300
Subject: [SR-Users] $avp reset to <null> with no reason?
In-Reply-To: <CAMMbDhQMMqM9GSGFjBGkH16N-wb6g=z0EfdYoy5AP59uYF-7Yg@mail.gmail.com>
References: <CAMMbDhQMMqM9GSGFjBGkH16N-wb6g=z0EfdYoy5AP59uYF-7Yg@mail.gmail.com>
Message-ID: <CAMMbDhRw9junf7nGrSTWk53twTxk8y6nOFfF8=Hp3AAw1Dog6Q@mail.gmail.com>

Sorry !!
I just notice that I used the avp once like "Sfeaturetype" and on the other
hand like "SfeatureType" (the "t" or "T"......)
Sorry again,
BR,
Uri

On Sun, Aug 12, 2012 at 2:50 PM, Uri Shacked <ushacked at gmail.com> wrote:

> Hi,****
>
> ** **
>
> I encountered a problem. While doing some avp's settings I notice that one
> avp that I use is changing to <null> without me doing anything.****
>
> ** **
>
> Here is the part of the cfg file:****
>
> ????.****
>
> ????.****
>
> Route[SET]{****
>
> ????..****
>
> $avp(Sfeaturetype)=$avp(next_feature_type);****
>
> ????****
>
> Route(DIDSRV);****
>
> }****
>
> ** **
>
> route[DIDSRV] {****
>
> xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV type = $avp(Sfeaturetype), index =
> $avp(SfeatureIndx) $C(xx)\n");****
>
> if($var(srvcount)>3){****
>
> xlog("L_NOTICE","$C(rg)Too many services loops $C(xx)\n");****
>
> update_stat("nts_subs_srv_loop", "+1");****
>
> $avp(TRMCS)="10";****
>
> t_reply(403,"too many loops");****
>
> exit;****
>
> }****
>
> $var(srvcount)=($var(srvcount)+1);****
>
> xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV before switch to
> $avp(SfeatureType) or $avp(next_feature_type) $C(xx)\n");****
>
> switch ($avp(SfeatureType)){****
>
> case "0" :****
>
> ???.****
>
> ???****
>
> ** **
>
> On the route "SET" I put the value from the $avp(next_feature_type) in
> $avp(Sfeaturetype).****
>
> After that I call the route "DIDSRV" and use switch($avp(Sfeaturetype)).
> Before the switch I xlog the value of the avp twice.****
>
> As you can see in the log below, on the first xlog print, the value is 0
> (as I expect). On the second print it resets to <null>****
>
> WHY?****
>
> ** **
>
> Log:****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:413]: qm_malloc(0x7fe6f9aff000, 72) returns address
> 0x7fe6fcc4d330 frag. 0x7fe6fcc4d300 (size=72) on 1 -th hit****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
> #033[0;31;42m SCRIPT: before DIDSRV type = 0, index = 8888888889
> #033[0;39;49m****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
> #033[0;31;42m SCRIPT*: in DIDSRV type = 0*, index = 8888888889 #033<8888888889%C2%A0%20%23033>
> [0;39;49m****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 848) called from <core>:
> rvalue.c: rval_new_empty(236)****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 848) returns address
> 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from
> <core>: rvalue.c: rval_destroy(141)****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from
> <core>: rvalue.c: rval_new_empty(236)****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
> #033[0;31;42m SCRIPT*: in DIDSRV before switch to <null>* or 0
> #033[0;39;49m****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 928) called from <core>:
> rvalue.c: rval_new_empty(236)****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 928) returns address
> 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from
> <core>: rvalue.c: rval_destroy(141)****
>
> Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from
> <core>: rvalue.c: rval_new_empty(236)
>
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From klaus.mailinglists at pernau.at  Mon Aug 13 11:37:10 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Mon, 13 Aug 2012 11:37:10 +0200
Subject: [SR-Users] Best tool for crafting sip messages
In-Reply-To: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
References: <CANBhyPASbEfo_K_1fsH7Sd93Je=qsXZdbXKwG46uiAahM0a88Q@mail.gmail.com>
Message-ID: <5028CAC6.7070400@pernau.at>

When I want to test special messages I write them in a test editor and 
send them with netcat.

When I want to test real call flows I often use a standard SIP client 
where I configure an outbound proxy. then on this outbound proxy 
(usually another Kamailio instance) I do message manipulation before 
forwarding the message.

regards
Klaus

On 13.08.2012 07:41, Anton Kvashenkin wrote:
> Hello, guys.
>
> What is the best tool for crafting sip messages for testing purpose? For
> example, simple REGISTER or INVITE message. I'm new to kamailio (not
> sip) so it would be great for me to use debbuger module for step-by-step
> digging into cfg.
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From apogrebennyk at sipwise.com  Mon Aug 13 11:53:03 2012
From: apogrebennyk at sipwise.com (Andrew Pogrebennyk)
Date: Mon, 13 Aug 2012 11:53:03 +0200
Subject: [SR-Users] Remove m=video from SDP header
In-Reply-To: <CAEprJzK3qJfU-_A-OPYjRq+iHnYjimTv-u3jrup7i0rDwxA4-g@mail.gmail.com>
References: <CAEprJzK3qJfU-_A-OPYjRq+iHnYjimTv-u3jrup7i0rDwxA4-g@mail.gmail.com>
Message-ID: <5028CE7F.1050203@sipwise.com>

You might need to upgrade to 3.3.1 to use sdpops module:
http://kamailio.org/docs/modules/stable/modules/sdpops.html

On 08/13/2012 09:52 AM, phillman25 wrote:
> Dear List
> 
> I am trying to remove specific lines from the following original SDP body:
> 
> 
> Content-Type: application/sdp
> Content-Length: 406
> 
> v=0
> o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:384
> t=0 0
> m=audio 11904 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 11602 RTP/AVP 34 99
> a=rtpmap:34 H263/90000
> a=rtpmap:99 H264/90000
> a=sendrecv
> 
> 
> 
> 
> I need to remove the following lines from the above SDP body as my
> International carrier does not support Video capabilities:
> 
> m=video 15042 RTP/AVP 34 99
> a=rtpmap:34 H263/90000
> a=rtpmap:99 H264/90000
> 
> 
> 
> 
> I used the following code in Kamailio config:
> 
> if(has_body("application/sdp") && search_body("m=video")){
> 
>                    subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #');
> 
>                    subst_body('#a=rtpmap:34 (.*)$# #');
> 
>                    subst_body('#a=rtpmap:99 (.*)$# #');
> 
>                    subst_body('#a=sendrecv(.*)$# #');
> 
>         }           
> 
> 
> 
> 
> Content-Type: application/sdp
> Content-Length: 325
> P-hint: outbound
> v=0
> o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:384
> t=0 0
> m=audio 11904 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> 
> As you can see the m=video body has been removed, however, calls are
> still failing. Is there something i have missed?
> 
> I am using Kamailio v3.2.2
> 
> 
> Thanking you in advance!
> 
> Phillip
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 



From govoiper at gmail.com  Mon Aug 13 15:33:59 2012
From: govoiper at gmail.com (SamyGo)
Date: Mon, 13 Aug 2012 18:33:59 +0500
Subject: [SR-Users] Kamailio Presence with XCAP not working accordingly
In-Reply-To: <CAJUJwtir5EpX0_a7Q_rB_G=mO6Sk2B1cuV4Ee6tP85hQpuGyiA@mail.gmail.com>
References: <CAJUJwth8p7pFaa6GVFER-B3SioZxPr2ZYF2Dxxewuz8N8kUHuA@mail.gmail.com>
	<1344587547.1881.5.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtg8hCc2w5jtNhvLRR5TSdn=VF-AyXK-Bi-4gBTrXhxfSQ@mail.gmail.com>
	<1344588814.1881.10.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtgUA-BpFfSfjMjdm-fcVcH97sz3x-_03sev-LW9vTLzJA@mail.gmail.com>
	<1344592828.1881.14.camel@pd-notebook-linux.croc.internal>
	<CAJUJwtgST=jfYQkBjLUb_ScJPND+SkXRooFfs+gXE9POPFkyWQ@mail.gmail.com>
	<CAJUJwtir5EpX0_a7Q_rB_G=mO6Sk2B1cuV4Ee6tP85hQpuGyiA@mail.gmail.com>
Message-ID: <CAJUJwthkKSBs4dfj2X3cETUR5NzbfuvQ1Yf-N3jHr3fTzkUUOg@mail.gmail.com>

A new discovery.!!  This configuration in the first email simply cool and
it works. But I think with kam-3.3 it doesn't !! I barely changed just one
module param option and it stops working but the same configuration file
works perfectly for 3.2 and 3.1 .

Though I will work with the 3.2 version but please someone test what needs
to be changed for newer version.

Thanks
BR
Sammy.
On Aug 10, 2012 3:41 PM, "SamyGo" <govoiper at gmail.com> wrote:

> Please see the SIP capture. As I was changing the online status from my
> jitis and eyebeam phone I could see the publish requests handled by
> kamailio but isn't it strange that I don'tfind any Notify generated and
> relayed to the watchers ?
>
>
> BR
> Sammy
>
>
> On Fri, Aug 10, 2012 at 3:04 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> Thats good enough for me, I'll go through everything and see if I can
>> make it work.
>> Thanks for your help and time.
>>
>>
>> On Fri, Aug 10, 2012 at 3:00 PM, Peter Dunkley <
>> peter.dunkley at crocodile-rcs.com> wrote:
>>
>>> **
>>> Hi,
>>>
>>> The only documentation I know of is the stuff in the tutorial, the RFCs,
>>> and the Kamailio module documentation.
>>>
>>> Offline chat is a SIP feature, it has nothing to do with the XCAP
>>> server.  The XCAP server manages XML documents (presence authorisation
>>> rules, contact lists, avatars, etc) and is accessed using HTTP.  Instant
>>> messages are SIP MESSAGE requests and are handled by the SIP routing part
>>> of Kamailio.
>>>
>>> If you want to do offline message handling you need the msilo Kamailio
>>> module (which isn't part of the tutorial).
>>>
>>> Regards,
>>>
>>> Peter
>>>
>>>
>>> On Fri, 2012-08-10 at 14:16 +0500, SamyGo wrote:
>>>
>>> Yes, Ekiga isn't good with it either. Can you just point me to some
>>> documentation and help me on that. I am sure I can get the presence
>>> working, but do you know hwo can I manage an offline chat thing in this !
>>> do I need a separate dedicated xcap server for this feature ?
>>>
>>>
>>>
>>>  On Fri, Aug 10, 2012 at 1:53 PM, Peter Dunkley <
>>> peter.dunkley at crocodile-rcs.com> wrote:
>>>
>>>  Hi,
>>>
>>> XCAP uses HTTP (albeit still carried on port 5060 in this tutorial),
>>> messaging and so on uses SIP.  These are different protocols and handled in
>>> kamailio.cfg differently.
>>>
>>> Last time I checked Xlite didn't support XCAP at all - so I don't think
>>> that's going to work at all.  I can't see XCAP/XDMS listed as an Ekiga
>>> feature either.
>>>
>>> I am using presence and XCAP on Kamailio 3.3 (my own configuration, not
>>> the one from the tutorial) and it works fine.
>>>
>>> Regards,
>>>
>>> Peter
>>>
>>>
>>>
>>> On Fri, 2012-08-10 at 13:41 +0500, SamyGo wrote:
>>>
>>> Thanks ,
>>>
>>>
>>> Yes this tutorial worked for me as well some 5/7 months ago with
>>> kamailio 3.1 but I'm on 3.3 now and tried the same configurations file.
>>> Jitsi on the other hand is the only phone I found which shows me this
>>> error. I'm trying with Ekiga - eyebeam or xlite didn't seem happy with the
>>> presence icons as well..But common thing was that I could send the Chat
>>> messages successfully.
>>>
>>>
>>> I'm continuously looking at the tcpdumps and I'll share those here in a
>>> while.
>>>
>>>
>>> Thanks for replying and taking interest.
>>>
>>>
>>> Regard,
>>> Sammy
>>>
>>>
>>> On Fri, Aug 10, 2012 at 1:32 PM, Peter Dunkley <
>>> peter.dunkley at crocodile-rcs.com> wrote:
>>>
>>> Hi,
>>>
>>> The xcap-caps document is hard-coded in kamailio.cfg and provides a list
>>> of the document types the XCAP server supports.  This part of the
>>> configuration from the tutorial builds and returns the xcap-caps document:
>>>
>>> 		if($xcapuri(u=>auid)=="xcap-caps")
>>> 		{
>>> 			$var(xbody) =
>>> "<?xml version='1.0' encoding='UTF-8'?>
>>> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'>
>>>   <auids>
>>>     <auid>rls-services</auid>
>>>     <auid>pidf-manipulation</auid>
>>>     <auid>xcap-caps</auid>
>>>     <auid>resource-lists</auid>
>>>     <auid>pres-rules</auid>
>>>     <auid>org.openmobilealliance.pres-rules</auid>
>>>   </auids>
>>>   <extensions>
>>>   </extensions>
>>>   <namespaces>
>>>     <namespace>urn:ietf:params:xml:ns:rls-services</namespace>
>>>     <namespace>urn:ietf:params:xml:ns:pidf</namespace>
>>>     <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace>
>>>     <namespace>urn:ietf:params:xml:ns:resource-lists</namespace>
>>>     <namespace>urn:ietf:params:xml:ns:pres-rules</namespace>
>>>   </namespaces>
>>> </xcap-caps>";
>>> 			xhttp_reply("200", "ok", "application/xcap-caps+xml",
>>> 					"$var(xbody)");
>>> 			exit;
>>> 		}
>>>
>>>  This tutorial has always worked for me in the past (although it has
>>> been well over a year since I last used it), so it looks like either a
>>> problem with Jitsi (which is unlikely) or some misconfiguration on the
>>> Kamailio side.
>>>
>>> A tcpdump of the traffic between Jitsi and Kamailio would help with
>>> working out which side has the problem.
>>>
>>> Regards,
>>>
>>> Peter
>>>
>>>
>>> On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
>>>
>>> Hi,
>>>
>>> I've followed the tutorial on kab.asipto.com for presence using
>>> built-in xcap server.
>>> http://kb.asipto.com/kamailio:presence:k31-made-simple
>>>
>>>
>>> I'm using Kamailio version 3.3.1 and did minor changes in modparams and
>>> rtpproxy function calls and the kamailio accepted the configurations file
>>> posted on the page and started.
>>>
>>>
>>> But the problem is that I don't get the presence status of the contacts
>>> still. Please suggest what to look for and how to troubleshoot this.
>>>
>>>
>>> I get this error on Jitsi - image attached.
>>>
>>>
>>> ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource
>>> can not be read"
>>>
>>>
>>> Regards,
>>> Sammy
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>   --
>>> Peter Dunkley
>>> Technical Director
>>> Crocodile RCS Ltd
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>   --
>>> Peter Dunkley
>>> Technical Director
>>> Crocodile RCS Ltd
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
>>>  _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>   --
>>> Peter Dunkley
>>> Technical Director
>>> Crocodile RCS Ltd
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>
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From agranig at sipwise.com  Mon Aug 13 23:04:25 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Mon, 13 Aug 2012 23:04:25 +0200
Subject: [SR-Users] Possible bad lcr module,
 after upgrade to 3.3.0 from 3.2
In-Reply-To: <20465.17444.614513.107789@siika.tutpro.com>
References: <CADJsLxgXLm-iPzZVVQQk=Lvqg69uk527uxeyQCWGZzCxc08rAQ@mail.gmail.com>
	<20450.49498.112879.142272@siika.tutpro.com>
	<CADJsLxhV0-MPBUGi9XtQi=f=H=9L5vAQcpHrwo7GMZUCPe5k+Q@mail.gmail.com>
	<20450.55305.266927.929348@siika.tutpro.com>
	<CADJsLxj1SV41WnwiLK7iwTLuUL1kBvAJOmdmaSPGKss6ez+geg@mail.gmail.com>
	<20450.60144.595521.82936@siika.tutpro.com>
	<CADJsLxjRTpnVnBvZHUGXOrvkbU5BryiS3+J_tHzZcT7hB5VfQQ@mail.gmail.com>
	<20450.64858.987000.752415@siika.tutpro.com>
	<CADJsLxhdQYCM7js9sfA7aZvUjSAnykf-dGuMqf4XB-pc3wF5-g@mail.gmail.com>
	<CADJsLxgYkB+6sYT9XvLiHqt2xHxEY-UXQSGCzeMkZqAVSonZYg@mail.gmail.com>
	<20465.17444.614513.107789@siika.tutpro.com>
Message-ID: <50296BD9.4090106@sipwise.com>

Hi,

Reviving this old thread, as we ran into the same issue as well. Turned
out it was a small glitch in passing a pointer to the size variable to
pcre, which was int instead of size_t, potentially causing stack
overflows on 64bit machines and messing with the rule_id, so the entry
got deactivated.

Richard has commited a fix today in trunk and 3.3, please try again to
see if it works now. If not, setting debug=5 and grepping for "lcr" in
the logs helps figuring out why the entry gets disabled.

Andreas

On 07/02/2012 08:48 AM, Juha Heinanen wrote:
> Alexey Mechanoshin writes:
> 
>> and if restart kamailio, again skipping from syslog:
>> skipping disabled <gw/rule> = <1/5>
>> skipping disabled <gw/rule> = <3/5>
> 
> i tried again with your lcr table records (enclosed) and i do not get those
> skipping messages.  i cannot do anything more unless i'm able to
> reproduce your problem.
> 
> -- juha
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 

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From jcastro at instant.com.br  Mon Aug 13 23:19:50 2012
From: jcastro at instant.com.br (Juan Castro)
Date: Mon, 13 Aug 2012 18:19:50 -0300
Subject: [SR-Users] How best to experiment with SIP over Websocket?
Message-ID: <CALRpQ=6mpV3_NuevA-SvRBQ1nwKtjSMyAair+2-ETL6dBDv9cg@mail.gmail.com>

As per the instructions here... http://www.kamailio.org/w/2012/07/websockets

...I downloaded the latest source from Git...
http://www.kamailio.org/w/download/

...but I don't see any mention of Websocket anywhere in the source. I
suspect I didn't actually download the development tree, but the 3.3
branch.

Last time I messed with any VoIP code was in 2008 (OpenSER and
Asterisk) and now I'm starting all over again. I apologize for the
dumb newbie question, but how do I test the Websocket code?

-- 
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regi?es: (11)4063-6100


From carsten at ng-voice.com  Mon Aug 13 23:21:54 2012
From: carsten at ng-voice.com (Carsten Bock)
Date: Mon, 13 Aug 2012 23:21:54 +0200
Subject: [SR-Users] How best to experiment with SIP over Websocket?
In-Reply-To: <CALRpQ=6mpV3_NuevA-SvRBQ1nwKtjSMyAair+2-ETL6dBDv9cg@mail.gmail.com>
References: <CALRpQ=6mpV3_NuevA-SvRBQ1nwKtjSMyAair+2-ETL6dBDv9cg@mail.gmail.com>
Message-ID: <CAOCjumFwJdDiPt5ZHa87DazUYT6bX=iEeZgnuY8t2sSrH5N49w@mail.gmail.com>

Hi,

Websocket support is only in the Trunk-Version of Kamailio, not in the
3.3 branch.

Carsten


2012/8/13 Juan Castro <jcastro at instant.com.br>:
> As per the instructions here... http://www.kamailio.org/w/2012/07/websockets
>
> ...I downloaded the latest source from Git...
> http://www.kamailio.org/w/download/
>
> ...but I don't see any mention of Websocket anywhere in the source. I
> suspect I didn't actually download the development tree, but the 3.3
> branch.



>
> Last time I messed with any VoIP code was in 2008 (OpenSER and
> Asterisk) and now I'm starting all over again. I apologize for the
> dumb newbie question, but how do I test the Websocket code?
>
> --
> Juan Carlos Castro y Castro
> Instant Solutions - Telefonia Gerando Resultado
> http://www.instant.com.br
> Principais capitais: 4063-6100
> Demais regi?es: (11)4063-6100
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
Carsten Bock
CEO (Gesch?ftsf?hrer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:carsten at ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Gesch?ftsf?hrer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/


From jcastro at instant.com.br  Tue Aug 14 00:06:59 2012
From: jcastro at instant.com.br (Juan Castro)
Date: Mon, 13 Aug 2012 19:06:59 -0300
Subject: [SR-Users] How best to experiment with SIP over Websocket?
In-Reply-To: <CAOCjumFwJdDiPt5ZHa87DazUYT6bX=iEeZgnuY8t2sSrH5N49w@mail.gmail.com>
References: <CALRpQ=6mpV3_NuevA-SvRBQ1nwKtjSMyAair+2-ETL6dBDv9cg@mail.gmail.com>
	<CAOCjumFwJdDiPt5ZHa87DazUYT6bX=iEeZgnuY8t2sSrH5N49w@mail.gmail.com>
Message-ID: <CALRpQ=7dsAzb25bdnojWy59AnT6q6rechBU-nRdLA+-fRGj5uA@mail.gmail.com>

Aha, compiled and installed, thanks. Now... what kind of SIP user
agent could I test this with?

On Mon, Aug 13, 2012 at 6:21 PM, Carsten Bock <carsten at ng-voice.com> wrote:
> Hi,
>
> Websocket support is only in the Trunk-Version of Kamailio, not in the
> 3.3 branch.
>
> Carsten
>
>
> 2012/8/13 Juan Castro <jcastro at instant.com.br>:
>> As per the instructions here... http://www.kamailio.org/w/2012/07/websockets
>>
>> ...I downloaded the latest source from Git...
>> http://www.kamailio.org/w/download/
>>
>> ...but I don't see any mention of Websocket anywhere in the source. I
>> suspect I didn't actually download the development tree, but the 3.3
>> branch.
>
>
>
>>
>> Last time I messed with any VoIP code was in 2008 (OpenSER and
>> Asterisk) and now I'm starting all over again. I apologize for the
>> dumb newbie question, but how do I test the Websocket code?
>>
>> --
>> Juan Carlos Castro y Castro
>> Instant Solutions - Telefonia Gerando Resultado
>> http://www.instant.com.br
>> Principais capitais: 4063-6100
>> Demais regi?es: (11)4063-6100
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Carsten Bock
> CEO (Gesch?ftsf?hrer)
>
> ng-voice GmbH
> Schomburgstr. 80
> D-22767 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:carsten at ng-voice.com
>
> Office +49 40 34927219
> Fax +49 40 34927220
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Gesch?ftsf?hrer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regi?es: (11)4063-6100


From carsten at ng-voice.com  Tue Aug 14 00:12:59 2012
From: carsten at ng-voice.com (Carsten Bock)
Date: Tue, 14 Aug 2012 00:12:59 +0200
Subject: [SR-Users] How best to experiment with SIP over Websocket?
In-Reply-To: <CALRpQ=7dsAzb25bdnojWy59AnT6q6rechBU-nRdLA+-fRGj5uA@mail.gmail.com>
References: <CALRpQ=6mpV3_NuevA-SvRBQ1nwKtjSMyAair+2-ETL6dBDv9cg@mail.gmail.com>
	<CAOCjumFwJdDiPt5ZHa87DazUYT6bX=iEeZgnuY8t2sSrH5N49w@mail.gmail.com>
	<CALRpQ=7dsAzb25bdnojWy59AnT6q6rechBU-nRdLA+-fRGj5uA@mail.gmail.com>
Message-ID: <CAOCjumH_q=kbnBp31hW0fzmrUtQBDLL3Ri78tYVd7C2EGgj6Mg@mail.gmail.com>

Please check this one:
http://www.sipml5.org

Don't use the online demo, since it will use a Websocket to SIP Gateway.

Carsten
Am 14.08.2012 00:07 schrieb "Juan Castro" <jcastro at instant.com.br>:

> Aha, compiled and installed, thanks. Now... what kind of SIP user
> agent could I test this with?
>
> On Mon, Aug 13, 2012 at 6:21 PM, Carsten Bock <carsten at ng-voice.com>
> wrote:
> > Hi,
> >
> > Websocket support is only in the Trunk-Version of Kamailio, not in the
> > 3.3 branch.
> >
> > Carsten
> >
> >
> > 2012/8/13 Juan Castro <jcastro at instant.com.br>:
> >> As per the instructions here...
> http://www.kamailio.org/w/2012/07/websockets
> >>
> >> ...I downloaded the latest source from Git...
> >> http://www.kamailio.org/w/download/
> >>
> >> ...but I don't see any mention of Websocket anywhere in the source. I
> >> suspect I didn't actually download the development tree, but the 3.3
> >> branch.
> >
> >
> >
> >>
> >> Last time I messed with any VoIP code was in 2008 (OpenSER and
> >> Asterisk) and now I'm starting all over again. I apologize for the
> >> dumb newbie question, but how do I test the Websocket code?
> >>
> >> --
> >> Juan Carlos Castro y Castro
> >> Instant Solutions - Telefonia Gerando Resultado
> >> http://www.instant.com.br
> >> Principais capitais: 4063-6100
> >> Demais regi?es: (11)4063-6100
> >>
> >> _______________________________________________
> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> >> sr-users at lists.sip-router.org
> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> >
> > --
> > Carsten Bock
> > CEO (Gesch?ftsf?hrer)
> >
> > ng-voice GmbH
> > Schomburgstr. 80
> > D-22767 Hamburg / Germany
> >
> > http://www.ng-voice.com
> > mailto:carsten at ng-voice.com
> >
> > Office +49 40 34927219
> > Fax +49 40 34927220
> >
> > Sitz der Gesellschaft: Hamburg
> > Registergericht: Amtsgericht Hamburg, HRB 120189
> > Gesch?ftsf?hrer: Carsten Bock
> > Ust-ID: DE279344284
> >
> > Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> > http://www.ng-voice.com/imprint/
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Juan Carlos Castro y Castro
> Instant Solutions - Telefonia Gerando Resultado
> http://www.instant.com.br
> Principais capitais: 4063-6100
> Demais regi?es: (11)4063-6100
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
-------------- next part --------------
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From Fchahrour at vanrise.com  Tue Aug 14 00:16:15 2012
From: Fchahrour at vanrise.com (Fatima Chahrour)
Date: Tue, 14 Aug 2012 01:16:15 +0300
Subject: [SR-Users] Kamailio and ACME
Message-ID: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>



Sent from iPhone


From jcastro at instant.com.br  Tue Aug 14 00:22:21 2012
From: jcastro at instant.com.br (Juan Castro)
Date: Mon, 13 Aug 2012 19:22:21 -0300
Subject: [SR-Users] How best to experiment with SIP over Websocket?
In-Reply-To: <CAOCjumH_q=kbnBp31hW0fzmrUtQBDLL3Ri78tYVd7C2EGgj6Mg@mail.gmail.com>
References: <CALRpQ=6mpV3_NuevA-SvRBQ1nwKtjSMyAair+2-ETL6dBDv9cg@mail.gmail.com>
	<CAOCjumFwJdDiPt5ZHa87DazUYT6bX=iEeZgnuY8t2sSrH5N49w@mail.gmail.com>
	<CALRpQ=7dsAzb25bdnojWy59AnT6q6rechBU-nRdLA+-fRGj5uA@mail.gmail.com>
	<CAOCjumH_q=kbnBp31hW0fzmrUtQBDLL3Ri78tYVd7C2EGgj6Mg@mail.gmail.com>
Message-ID: <CALRpQ=5aU0bvzueGN7yeQoY9=oPTa=zTW6ESfnT-52-=Qbxuug@mail.gmail.com>

So, if I leech that code (the source download is exactly the same as
their site), host it in an Apache server of mine, change the hardcoded
gateway address in the demo to mine, I have a chance of success?

Sounds cool!

On Mon, Aug 13, 2012 at 7:12 PM, Carsten Bock <carsten at ng-voice.com> wrote:
> Please check this one:
> http://www.sipml5.org
>
> Don't use the online demo, since it will use a Websocket to SIP Gateway.
>
> Carsten
>
> Am 14.08.2012 00:07 schrieb "Juan Castro" <jcastro at instant.com.br>:
>
>> Aha, compiled and installed, thanks. Now... what kind of SIP user
>> agent could I test this with?
>>
>> On Mon, Aug 13, 2012 at 6:21 PM, Carsten Bock <carsten at ng-voice.com>
>> wrote:
>> > Hi,
>> >
>> > Websocket support is only in the Trunk-Version of Kamailio, not in the
>> > 3.3 branch.
>> >
>> > Carsten
>> >
>> >
>> > 2012/8/13 Juan Castro <jcastro at instant.com.br>:
>> >> As per the instructions here...
>> >> http://www.kamailio.org/w/2012/07/websockets
>> >>
>> >> ...I downloaded the latest source from Git...
>> >> http://www.kamailio.org/w/download/
>> >>
>> >> ...but I don't see any mention of Websocket anywhere in the source. I
>> >> suspect I didn't actually download the development tree, but the 3.3
>> >> branch.
>> >
>> >
>> >
>> >>
>> >> Last time I messed with any VoIP code was in 2008 (OpenSER and
>> >> Asterisk) and now I'm starting all over again. I apologize for the
>> >> dumb newbie question, but how do I test the Websocket code?
>> >>
>> >> --
>> >> Juan Carlos Castro y Castro
>> >> Instant Solutions - Telefonia Gerando Resultado
>> >> http://www.instant.com.br
>> >> Principais capitais: 4063-6100
>> >> Demais regi?es: (11)4063-6100
>> >>
>> >> _______________________________________________
>> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> >> sr-users at lists.sip-router.org
>> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >
>> >
>> >
>> > --
>> > Carsten Bock
>> > CEO (Gesch?ftsf?hrer)
>> >
>> > ng-voice GmbH
>> > Schomburgstr. 80
>> > D-22767 Hamburg / Germany
>> >
>> > http://www.ng-voice.com
>> > mailto:carsten at ng-voice.com
>> >
>> > Office +49 40 34927219
>> > Fax +49 40 34927220
>> >
>> > Sitz der Gesellschaft: Hamburg
>> > Registergericht: Amtsgericht Hamburg, HRB 120189
>> > Gesch?ftsf?hrer: Carsten Bock
>> > Ust-ID: DE279344284
>> >
>> > Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>> > http://www.ng-voice.com/imprint/
>> >
>> > _______________________________________________
>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> > sr-users at lists.sip-router.org
>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> --
>> Juan Carlos Castro y Castro
>> Instant Solutions - Telefonia Gerando Resultado
>> http://www.instant.com.br
>> Principais capitais: 4063-6100
>> Demais regi?es: (11)4063-6100
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regi?es: (11)4063-6100


From abalashov at evaristesys.com  Tue Aug 14 00:44:42 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 13 Aug 2012 18:44:42 -0400
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>
Message-ID: <5029835A.4000108@evaristesys.com>

Is there a predicate?

On 08/13/2012 06:16 PM, Fatima Chahrour wrote:

>
>
> Sent from iPhone
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From ronh at netweave.com  Tue Aug 14 01:11:33 2012
From: ronh at netweave.com (Ron Hopson)
Date: Mon, 13 Aug 2012 17:11:33 -0600
Subject: [SR-Users] UAC module
Message-ID: <502989A5.8000908@netweave.com>

Hello,

I cannot create a configuration consistent with uac_reg_lookup. Does 
someone have a configuration with a simple example of uac_reg_lookup 
that works with db_text file uacreg?

I have kamailio 3.3 running on linux.  Based on suggestions in
http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html
and
http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html
I created this record in db_text file uacreg:
2:12345678:daniel:mysipserver.com:daniel123:sipprovider.com:sipprovider.com:daniel123:xxxxx:"sip:sipprovider.com"

I assume that function uac_reg_lookup retrieves registration information 
from this record after it is loaded into a memory table.
One version of the configuration file is attached.  Here are three 
examples of failures to create a configuration.

     $rU = 12345678;

     if(uac_reg_lookup("$rU", "$ru")){            #line 73

  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 1
  0(13856) DEBUG: <core> [rvalue.c:2861]: RV is int: 12345678
  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
  0(13856) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
  0(13856) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
cfg:/usr/local/etc/kamailio/kamailio.cfg:73
---------
     $rU = "12345678";

     if(uac_reg_lookup("$rU", "$ru")){            #line 73

  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
  0(13973) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
  0(13973) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
  0(13973) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
cfg:/usr/local/etc/kamailio/kamailio.cfg:73
--------
     $rU = "12345678";
     $ru = "sip:mysipserver.com";

     if(uac_reg_lookup("$rU", "$ru")){            #line 74


  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "sip:mysipserver.com"
  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
  0(13976) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
  0(13976) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
cfg:/usr/local/etc/kamailio/kamailio.cfg:74

---------------------
regards,
Ron Hopson

-------------- next part --------------
#
# $Id$
#
# simple quick-start config script with dbtext
#

# ----------- global configuration parameters ------------------------

debug=9         # debug level (cmd line: -dddddddddd)
#fork=yes
log_stderror=yes
log_facility=LOG_LOCAL2

check_via=no    # (cmd. line: -v)
dns=no          # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
children=4
disable_tcp=yes

listen=173.63.113.62
port=5060

#!define DBURL "text:///home/agiletel/kamailio/dbtext"

# ------------------ module loading ----------------------------------
mpath="/usr/local/lib/kamailio/"

# use dbtext database
loadmodule "modules_k/db_text.so"

loadmodule "modules/debugger.so"
loadmodule "modules_k/mi_fifo.so"
loadmodule "modules/tm.so"
loadmodule "modules/sl.so"
loadmodule "modules_k/rr.so"
loadmodule "modules_k/maxfwd.so"
loadmodule "modules_k/usrloc.so"
loadmodule "modules_k/registrar.so"
loadmodule "modules_k/textops.so"

loadmodule "modules_k/uac.so"
loadmodule "modules_k/xlog.so"


# ----------------- setting module-specific parameters ---------------

# -- mi_fifo params --

modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")

# -- usrloc params --

# use dbtext database for persistent storage
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", DBURL)

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# -- uacreg params --
modparam("uac","restore_mode","auto")


# -------------------------  request routing logic -------------------

# main routing logic

route{

	$rU = "12345678";
	$ru = "sip:mysipserver.com";

	if(uac_reg_lookup("$rU", "$ru")){
		xlog("uac_reg_lookup found remote user [$rU] on [$rd] via [$du]\n");
	    lookup("location");
	}

}

From miconda at gmail.com  Tue Aug 14 08:59:57 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 08:59:57 +0200
Subject: [SR-Users] UAC module
In-Reply-To: <502989A5.8000908@netweave.com>
References: <502989A5.8000908@netweave.com>
Message-ID: <5029F76D.6090903@gmail.com>

Hello,

the parameters are expected to be variables, but looking at the log 
messages it seems that config optimization replace them with the string 
values, at you have the assignment just above the function, the 
parameters will be those string values always at runtime.

Can you remove the assign statements and try again to see if starts ok?

Cheers,
Daniel

On 8/14/12 1:11 AM, Ron Hopson wrote:
> Hello,
>
> I cannot create a configuration consistent with uac_reg_lookup. Does 
> someone have a configuration with a simple example of uac_reg_lookup 
> that works with db_text file uacreg?
>
> I have kamailio 3.3 running on linux.  Based on suggestions in
> http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html
> and
> http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html 
>
> I created this record in db_text file uacreg:
> 2:12345678:daniel:mysipserver.com:daniel123:sipprovider.com:sipprovider.com:daniel123:xxxxx:"sip:sipprovider.com" 
>
>
> I assume that function uac_reg_lookup retrieves registration 
> information from this record after it is loaded into a memory table.
> One version of the configuration file is attached.  Here are three 
> examples of failures to create a configuration.
>
>     $rU = 12345678;
>
>     if(uac_reg_lookup("$rU", "$ru")){            #line 73
>
>  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 1
>  0(13856) DEBUG: <core> [rvalue.c:2861]: RV is int: 12345678
>  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>  0(13856) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>  0(13856) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
> cfg:/usr/local/etc/kamailio/kamailio.cfg:73
> ---------
>     $rU = "12345678";
>
>     if(uac_reg_lookup("$rU", "$ru")){            #line 73
>
>  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>  0(13973) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
>  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>  0(13973) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>  0(13973) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
> cfg:/usr/local/etc/kamailio/kamailio.cfg:73
> --------
>     $rU = "12345678";
>     $ru = "sip:mysipserver.com";
>
>     if(uac_reg_lookup("$rU", "$ru")){            #line 74
>
>
>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "sip:mysipserver.com"
>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>  0(13976) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>  0(13976) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
> cfg:/usr/local/etc/kamailio/kamailio.cfg:74
>
> ---------------------
> regards,
> Ron Hopson
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From miconda at gmail.com  Tue Aug 14 11:15:34 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 11:15:34 +0200
Subject: [SR-Users] Remove m=video from SDP header
In-Reply-To: <CAEprJzK3qJfU-_A-OPYjRq+iHnYjimTv-u3jrup7i0rDwxA4-g@mail.gmail.com>
References: <CAEprJzK3qJfU-_A-OPYjRq+iHnYjimTv-u3jrup7i0rDwxA4-g@mail.gmail.com>
Message-ID: <502A1736.7070106@gmail.com>

Hello,

the sdp looks ok, assuming that there is an empty line after P-hint header.

Perhaps you should test with a sip phone to see if the call is ok with 
it, if yes, then you have to ask the carrier to give you details about 
why the call is not accepted.

Cheers,
DAniel

On 8/13/12 9:52 AM, phillman25 wrote:
> Dear List
>
> I am trying to remove specific lines from the following original SDP body:
>
>
> Content-Type: application/sdp
> Content-Length: 406
>
> v=0
> o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:384
> t=0 0
> m=audio 11904 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 11602 RTP/AVP 34 99
> a=rtpmap:34 H263/90000
> a=rtpmap:99 H264/90000
> a=sendrecv
>
>
>
>
> I need to remove the following lines from the above SDP body as my 
> International carrier does not support Video capabilities:
>
> m=video 15042 RTP/AVP 34 99
> a=rtpmap:34 H263/90000
> a=rtpmap:99 H264/90000
>
>
>
>
> I used the following code in Kamailio config:
>
> if(has_body("application/sdp") && search_body("m=video")){
>
>  subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #');
>
>    subst_body('#a=rtpmap:34 (.*)$# #');
>
>  subst_body('#a=rtpmap:99 (.*)$# #');
>
>  subst_body('#a=sendrecv(.*)$# #');
>
>         }
>
>
>
>
> Content-Type: application/sdp
> Content-Length: 325
> P-hint: outbound
> v=0
> o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:384
> t=0 0
> m=audio 11904 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
> As you can see the m=video body has been removed, however, calls are 
> still failing. Is there something i have missed?
>
> I am using Kamailio v3.2.2
>
>
> Thanking you in advance!
>
> Phillip

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From miconda at gmail.com  Tue Aug 14 11:21:56 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 11:21:56 +0200
Subject: [SR-Users] centos (fedora) service startup file
In-Reply-To: <501AA9D6.4040500@rodnoe.tv>
References: <501AA9D6.4040500@rodnoe.tv>
Message-ID: <502A18B4.8040207@gmail.com>

Hello,

thanks, it was committed in git master branch, backports to follow 
before minor releases.

Do you build your own rpms or you install from the opensuse build system?

Cheers,
Daniel

On 8/2/12 6:24 PM, andre wrote:
> Somwhere losted since openser 1.0 :)
>
>
> Please fix in:
>
> start() {
>         echo -n $"Starting $prog: "
> .....
>    [ $RETVAL = 0 ] && touch something
> to
>   [ $RETVAL = 0 ] && touch someting && success
>
> I mean call function
> success
> in the end of the expression.
>
> Thanks in advance
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From fchahrour at vanrise.com  Tue Aug 14 11:29:49 2012
From: fchahrour at vanrise.com (Fatima Chahrour~Vanrise Support)
Date: Tue, 14 Aug 2012 12:29:49 +0300
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <5029835A.4000108@evaristesys.com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>
	<5029835A.4000108@evaristesys.com>
Message-ID: <065801cd79ff$5b491530$11db3f90$@com>

Hi,

Sorry for sending this incompletely.
I wanted to ask if Kamailio is compatible with ACME Packet or experianced by
anyone in the list?
and if there is any document that you recommend about such topic espacially
in what is related to routing?

Is it like having kamailio with any SBC?

Thanks in advance,

F.Chahrour

-----Original Message-----
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Tuesday, August 14, 2012 1:45 AM
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Kamailio and ACME

Is there a predicate?

On 08/13/2012 06:16 PM, Fatima Chahrour wrote:

>
>
> Sent from iPhone
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



From agranig at sipwise.com  Tue Aug 14 11:47:11 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 14 Aug 2012 11:47:11 +0200
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <065801cd79ff$5b491530$11db3f90$@com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>
	<5029835A.4000108@evaristesys.com>
	<065801cd79ff$5b491530$11db3f90$@com>
Message-ID: <502A1E9F.5070608@sipwise.com>

On 08/14/2012 11:29 AM, Fatima Chahrour~Vanrise Support wrote:
> I wanted to ask if Kamailio is compatible with ACME Packet or experianced by
> anyone in the list?
> and if there is any document that you recommend about such topic espacially
> in what is related to routing?

There is nothing special for interoperating Kamailio with ACME, it's
treated like any other SIP device. As always, it depends on what you
want to do.

Andreas

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From fchahrour at vanrise.com  Tue Aug 14 12:04:10 2012
From: fchahrour at vanrise.com (Fatima Chahrour~Vanrise Support)
Date: Tue, 14 Aug 2012 13:04:10 +0300
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <502A1E9F.5070608@sipwise.com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>	<5029835A.4000108@evaristesys.com>	<065801cd79ff$5b491530$11db3f90$@com>
	<502A1E9F.5070608@sipwise.com>
Message-ID: <067701cd7a04$27fa37e0$77eea7a0$@com>

Thank you Andrea. 
I just wanted to check if there is any incompatibility between both before
surfing deeply.

-----Original Message-----
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Andreas Granig
Sent: Tuesday, August 14, 2012 12:47 PM
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Kamailio and ACME

On 08/14/2012 11:29 AM, Fatima Chahrour~Vanrise Support wrote:
> I wanted to ask if Kamailio is compatible with ACME Packet or 
> experianced by anyone in the list?
> and if there is any document that you recommend about such topic 
> espacially in what is related to routing?

There is nothing special for interoperating Kamailio with ACME, it's treated
like any other SIP device. As always, it depends on what you want to do.

Andreas




From bligthart at btlnet.co.uk  Tue Aug 14 12:05:50 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Tue, 14 Aug 2012 10:05:50 +0000
Subject: [SR-Users] db_cassandra
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>

Hi,

I am trying to get the db_cassandra working in kamailio 3.1.1 with thrift 0.6.1 API and however I get the following
compile error.

ol.h:33,
                 from dbcassa_base.h:38,
                 from dbcassa_base.cpp:47:
../../lib/srdb1/db.h:388:20: error: expected unqualified-id before ?delete?
dbcassa_base.cpp: In function ?ColumnVecPtr cassa_translate_query(const db1_con_t*, str* const*, const db_val_t*, str* const*, int, int, int*)?:
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 7 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 5 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 6 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 4 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 4 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
../../lib/srdb1/db_res.h: At global scope:
../../lib/srdb1/db_res.h:105:19: warning: inline function ?db1_res_t* db_new_result()? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:121:12: warning: inline function ?int db_allocate_columns(db1_res_t*, unsigned int)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:98:12: warning: inline function ?int db_free_columns(db1_res_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:129:12: warning: inline function ?int db_allocate_rows(db1_res_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_row.h:74:12: warning: inline function ?int db_allocate_row(const db1_res_t*, db_row_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:111:12: warning: inline function ?int db_free_result(db1_res_t*)? used but never defined [enabled by default]
make[1]: *** [dbcassa_base.o] Error 1
make: *** [install-modules] Error 1


Anybody has got any ideas?

Thanks in advance,

Bou

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From phillman25 at gmail.com  Tue Aug 14 13:55:49 2012
From: phillman25 at gmail.com (phillman25)
Date: Tue, 14 Aug 2012 14:55:49 +0300
Subject: [SR-Users] Remove m=video from SDP header
In-Reply-To: <502A1736.7070106@gmail.com>
References: <CAEprJzK3qJfU-_A-OPYjRq+iHnYjimTv-u3jrup7i0rDwxA4-g@mail.gmail.com>
	<502A1736.7070106@gmail.com>
Message-ID: <CAEprJzJqHHmXkFtQrMuyCpayCKE5FP3K+=6SfrjGXj0Q1bZSiQ@mail.gmail.com>

Hi Daniel

Yes there is an empty line after P-hint header.
I will make a few tests as you suggested below to see whether its the
carrier.

Thanks
Phillip

On Tue, Aug 14, 2012 at 12:15 PM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

>  Hello,
>
> the sdp looks ok, assuming that there is an empty line after P-hint header.
>
> Perhaps you should test with a sip phone to see if the call is ok with it,
> if yes, then you have to ask the carrier to give you details about why the
> call is not accepted.
>
> Cheers,
> DAniel
>
>
> On 8/13/12 9:52 AM, phillman25 wrote:
>
> Dear List
>
>  I am trying to remove specific lines from the following original SDP
> body:
>
>
>  Content-Type: application/sdp
> Content-Length: 406
>
>  v=0
> o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:384
> t=0 0
> m=audio 11904 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>  a=sendrecv
> m=video 11602 RTP/AVP 34 99
> a=rtpmap:34 H263/90000
> a=rtpmap:99 H264/90000
> a=sendrecv
>
>
>
>
>  I need to remove the following lines from the above SDP body as my
> International carrier does not support Video capabilities:
>
> m=video 15042 RTP/AVP 34 99
> a=rtpmap:34 H263/90000
> a=rtpmap:99 H264/90000
>
>
>
>  I used the following code in Kamailio config:
>
> if(has_body("application/sdp") && search_body("m=video")){
>
>                    subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #');
>
>                    subst_body('#a=rtpmap:34 (.*)$# #');
>
>                    subst_body('#a=rtpmap:99 (.*)$# #');
>
>                    subst_body('#a=sendrecv(.*)$# #');
>
>         }
>
>
>
>  Content-Type: application/sdp
> Content-Length: 325
> P-hint: outbound
> v=0
> o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
>  s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> b=CT:384
> t=0 0
> m=audio 11904 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
>  As you can see the m=video body has been removed, however, calls are
> still failing. Is there something i have missed?
>
>  I am using Kamailio v3.2.2
>
>
>  Thanking you in advance!
>
>  Phillip
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From fchahrour at vanrise.com  Tue Aug 14 14:25:17 2012
From: fchahrour at vanrise.com (Fatima Chahrour~Vanrise Support)
Date: Tue, 14 Aug 2012 15:25:17 +0300
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <067701cd7a04$27fa37e0$77eea7a0$@com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>	<5029835A.4000108@evaristesys.com>	<065801cd79ff$5b491530$11db3f90$@com>	<502A1E9F.5070608@sipwise.com>
	<067701cd7a04$27fa37e0$77eea7a0$@com>
Message-ID: <06b501cd7a17$df1452e0$9d3cf8a0$@com>

Can I ask one more question please?!

Am planning to send the routing decision configured in kamailio to ACME SBC;
but wanted to know if anyone has an idea about the format of this routing
that ACME will take from Kamailio. In other words, what is the routing
output from Kamailio and that works to be as an input for ACME? And how will
ACME forward a call to Kamailio to get this routing?

For example let's take the following example which summarize my target:
Call from a customer will hit the ACME SBC, and the ACME SBC which will take
the routing choice from Kamailio. 


1- EX call to 9613555555 with Supplier A as first choice in routing and
Supplier B as second choice with less priority (on Kamailio)
2- A customer on ACME calls 9613555555 where ACME will send request to
Kamailio and get supplier A as result
3- ACME will try the call on supplier B (either succeeds or fails)
4- In case of failure, ACME should return (something!!!) to Kamailio and get
Supplier B as second choice.

Can you explain how the above is configures between ACME (or any other SBC)
and Kamailio?

Regards,
F.Chahrour

-----Original Message-----
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fatima
Chahrour~Vanrise Support
Sent: Tuesday, August 14, 2012 1:04 PM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Re: [SR-Users] Kamailio and ACME

Thank you Andrea. 
I just wanted to check if there is any incompatibility between both before
surfing deeply.

-----Original Message-----
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Andreas Granig
Sent: Tuesday, August 14, 2012 12:47 PM
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Kamailio and ACME

On 08/14/2012 11:29 AM, Fatima Chahrour~Vanrise Support wrote:
> I wanted to ask if Kamailio is compatible with ACME Packet or 
> experianced by anyone in the list?
> and if there is any document that you recommend about such topic 
> espacially in what is related to routing?

There is nothing special for interoperating Kamailio with ACME, it's treated
like any other SIP device. As always, it depends on what you want to do.

Andreas



_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



From anca.vamanu at 1and1.ro  Tue Aug 14 14:39:26 2012
From: anca.vamanu at 1and1.ro (Anca Vamanu)
Date: Tue, 14 Aug 2012 15:39:26 +0300
Subject: [SR-Users] db_cassandra
In-Reply-To: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
Message-ID: <502A46FE.7080403@1and1.ro>

Hi,


I tried compiling the module with the latest 3.1 from git and indeed it 
did not compile initially.
This very simple patch that is attached was needed. There were some 
problems with implicit casts.

I see that you are talking about 3.1.1 version, I have not tried with 
this exact version. You should maybe consider using the latest git for 
3.1, it should also be more stable.


Regards,
Anca





On 08/14/2012 01:05 PM, Boudewyn Ligthart wrote:
>
> Hi,
>
> I am trying to get the db_cassandra working in kamailio 3.1.1 with 
> thrift 0.6.1 API and however I get the following
> compile error.
>
> /ol.h:33,/
>
> /                 from dbcassa_base.h:38,/
>
> /                 from dbcassa_base.cpp:47:/
>
> /../../lib/srdb1/db.h:388:20: error: expected unqualified-id before 
> ?delete?/
>
> /dbcassa_base.cpp: In function ?ColumnVecPtr 
> cassa_translate_query(const db1_con_t*, str* const*, const db_val_t*, 
> str* const*, int, int, int*)?:/
>
> /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type 
> ?int?, but argument 7 has type 
> ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long 
> unsigned int}? [-Wformat]/
>
> /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type 
> ?int?, but argument 5 has type 
> ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long 
> unsigned int}? [-Wformat]/
>
> /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type 
> ?int?, but argument 6 has type 
> ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long 
> unsigned int}? [-Wformat]/
>
> /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type 
> ?int?, but argument 4 has type 
> ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long 
> unsigned int}? [-Wformat]/
>
> /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type 
> ?int?, but argument 4 has type 
> ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long 
> unsigned int}? [-Wformat]/
>
> /../../lib/srdb1/db_res.h: At global scope:/
>
> /../../lib/srdb1/db_res.h:105:19: warning: inline function ?db1_res_t* 
> db_new_result()? used but never defined [enabled by default]/
>
> /../../lib/srdb1/db_res.h:121:12: warning: inline function ?int 
> db_allocate_columns(db1_res_t*, unsigned int)? used but never defined 
> [enabled by default]/
>
> /../../lib/srdb1/db_res.h:98:12: warning: inline function ?int 
> db_free_columns(db1_res_t*)? used but never defined [enabled by default]/
>
> /../../lib/srdb1/db_res.h:129:12: warning: inline function ?int 
> db_allocate_rows(db1_res_t*)? used but never defined [enabled by default]/
>
> /../../lib/srdb1/db_row.h:74:12: warning: inline function ?int 
> db_allocate_row(const db1_res_t*, db_row_t*)? used but never defined 
> [enabled by default]/
>
> /../../lib/srdb1/db_res.h:111:12: warning: inline function ?int 
> db_free_result(db1_res_t*)? used but never defined [enabled by default]/
>
> /make[1]: *** [dbcassa_base.o] Error 1/
>
> /make: *** [install-modules] Error 1/
>
> //
>
> //
>
> Anybody has got any ideas?
>
> Thanks in advance,
>
> Bou
>


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From phillman25 at gmail.com  Tue Aug 14 14:47:03 2012
From: phillman25 at gmail.com (phillman25)
Date: Tue, 14 Aug 2012 15:47:03 +0300
Subject: [SR-Users] rtpproxy module issue
Message-ID: <CAEprJzKhaveUa-b_cPOP+V8kkLo97N+CERmdnyd94eCiJuhJPA@mail.gmail.com>

Dear List

I have just installed kamailio V 3.3.1
However, when trying to start kamailio its giving me the below error:


ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module
</usr/local/lib/kamailio/modules_k/rtpproxy.so>:
/usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc
: <core> [cfg.y:3591]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
load module


Does anyone know why rtpproxy module cannot be loaded?


Thanks
Phillip
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From abalashov at evaristesys.com  Tue Aug 14 14:49:13 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Tue, 14 Aug 2012 08:49:13 -0400
Subject: [SR-Users] rtpproxy module issue
In-Reply-To: <CAEprJzKhaveUa-b_cPOP+V8kkLo97N+CERmdnyd94eCiJuhJPA@mail.gmail.com>
References: <CAEprJzKhaveUa-b_cPOP+V8kkLo97N+CERmdnyd94eCiJuhJPA@mail.gmail.com>
Message-ID: <502A4949.2000707@evaristesys.com>

Please post output of:

ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so

On 08/14/2012 08:47 AM, phillman25 wrote:

> Dear List
>
> I have just installed kamailio V 3.3.1
> However, when trying to start kamailio its giving me the below error:
>
>
> ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open
> module </usr/local/lib/kamailio/modules_k/rtpproxy.so>:
> /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc
> : <core> [cfg.y:3591]: parse error in config file
> /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
> load module
>
>
> Does anyone know why rtpproxy module cannot be loaded?
>
>
> Thanks
> Phillip
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From bligthart at btlnet.co.uk  Tue Aug 14 15:26:59 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Tue, 14 Aug 2012 13:26:59 +0000
Subject: [SR-Users] db_cassandra
In-Reply-To: <502A46FE.7080403@1and1.ro>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A46FE.7080403@1and1.ro>
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11AF60@DB3PRD0510MB380.eurprd05.prod.outlook.com>

HI Anca,

Thanks for pointing me to the latest GIT version, this compiles as per normal.

Another question,

Have you got pointers on how to install the tables / setup Cassandra.
I have created the column families for location, subscriber, version but not yet filled these with data.

Regards

Bou

From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Anca Vamanu
Sent: 14 August 2012 13:39
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] db_cassandra

Hi,


I tried compiling the module with the latest 3.1 from git and indeed it did not compile initially.
This very simple patch that is attached was needed. There were some problems with implicit casts.

I see that you are talking about 3.1.1 version, I have not tried with this exact version. You should maybe consider using the latest git for 3.1, it should also be more stable.


Regards,
Anca





On 08/14/2012 01:05 PM, Boudewyn Ligthart wrote:
Hi,

I am trying to get the db_cassandra working in kamailio 3.1.1 with thrift 0.6.1 API and however I get the following
compile error.

ol.h:33,
                 from dbcassa_base.h:38,
                 from dbcassa_base.cpp:47:
../../lib/srdb1/db.h:388:20: error: expected unqualified-id before ?delete?
dbcassa_base.cpp: In function ?ColumnVecPtr cassa_translate_query(const db1_con_t*, str* const*, const db_val_t*, str* const*, int, int, int*)?:
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 7 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 5 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 6 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 4 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 4 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
../../lib/srdb1/db_res.h: At global scope:
../../lib/srdb1/db_res.h:105:19: warning: inline function ?db1_res_t* db_new_result()? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:121:12: warning: inline function ?int db_allocate_columns(db1_res_t*, unsigned int)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:98:12: warning: inline function ?int db_free_columns(db1_res_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:129:12: warning: inline function ?int db_allocate_rows(db1_res_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_row.h:74:12: warning: inline function ?int db_allocate_row(const db1_res_t*, db_row_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:111:12: warning: inline function ?int db_free_result(db1_res_t*)? used but never defined [enabled by default]
make[1]: *** [dbcassa_base.o] Error 1
make: *** [install-modules] Error 1


Anybody has got any ideas?

Thanks in advance,

Bou


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From phillman25 at gmail.com  Tue Aug 14 15:50:42 2012
From: phillman25 at gmail.com (phillman25)
Date: Tue, 14 Aug 2012 16:50:42 +0300
Subject: [SR-Users] rtpproxy module issue
Message-ID: <CAEprJz+edBEy4pPVmHaqRG7YRbW7URS8KayqARSaoK1Lswhq8A@mail.gmail.com>

Hello

This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so:



shell-init: error retrieving current directory: getcwd: cannot access
parent directories: No such file or directory
        linux-gate.so.1 =>  (0x00276000)
        libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000)
        libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000)
        libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000)
        /lib/ld-linux.so.2 (0x00114000)



Regards
Phillip





Message: 3
Date: Tue, 14 Aug 2012 08:49:13 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [SR-Users] rtpproxy module issue
To: sr-users at lists.sip-router.org
Message-ID: <502A4949.2000707 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Please post output of:

ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so

On 08/14/2012 08:47 AM, phillman25 wrote:

> Dear List
>
> I have just installed kamailio V 3.3.1
> However, when trying to start kamailio its giving me the below error:
>
>
> ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open
> module </usr/local/lib/kamailio/modules_k/rtpproxy.so>:
> /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol:
qm_realloc
> : <core> [cfg.y:3591]: parse error in config file
> /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
> load module
>
>
> Does anyone know why rtpproxy module cannot be loaded?
>
>
> Thanks
> Phillip
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
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From anca.vamanu at 1and1.ro  Tue Aug 14 15:52:41 2012
From: anca.vamanu at 1and1.ro (Anca Vamanu)
Date: Tue, 14 Aug 2012 16:52:41 +0300
Subject: [SR-Users] db_cassandra
In-Reply-To: <D8261FE064C78B44B77EF7C4569D6CBF0D11AF60@DB3PRD0510MB380.eurprd05.prod.outlook.com>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A46FE.7080403@1and1.ro>
	<D8261FE064C78B44B77EF7C4569D6CBF0D11AF60@DB3PRD0510MB380.eurprd05.prod.outlook.com>
Message-ID: <502A5829.3090201@1and1.ro>

Hi,


If you are using cassandra-cli you can fill in the tables using 
something like this:

version table:

set version['location']['table_version']='1004';
set version['subscriber']['table_version']='6';

and the subscriber table:

set subscriber['userX']['domain']='domain.com';
set subscriber['userX']['password']='pass123';


Regards,
Anca


On 08/14/2012 04:26 PM, Boudewyn Ligthart wrote:
>
> HI Anca,
>
> Thanks for pointing me to the latest GIT version, this compiles as per 
> normal.
>
> Another question,
>
> Have you got pointers on how to install the tables / setup Cassandra.
>
> I have created the column families for location, subscriber, version 
> but not yet filled these with data.
>
> Regards
>
> Bou
>
> *From:*sr-users-bounces at lists.sip-router.org 
> [mailto:sr-users-bounces at lists.sip-router.org] *On Behalf Of *Anca Vamanu
> *Sent:* 14 August 2012 13:39
> *To:* sr-users at lists.sip-router.org
> *Subject:* Re: [SR-Users] db_cassandra
>
> Hi,
>
>
> I tried compiling the module with the latest 3.1 from git and indeed 
> it did not compile initially.
> This very simple patch that is attached was needed. There were some 
> problems with implicit casts.
>
> I see that you are talking about 3.1.1 version, I have not tried with 
> this exact version. You should maybe consider using the latest git for 
> 3.1, it should also be more stable.
>
>
> Regards,
> Anca
>
>
>
>
>
> On 08/14/2012 01:05 PM, Boudewyn Ligthart wrote:
>
>     Hi,
>
>     I am trying to get the db_cassandra working in kamailio 3.1.1 with
>     thrift 0.6.1 API and however I get the following
>     compile error.
>
>     /ol.h:33,/
>
>     /                 from dbcassa_base.h:38,/
>
>     /                 from dbcassa_base.cpp:47:/
>
>     /../../lib/srdb1/db.h:388:20: error: expected unqualified-id
>     before ?delete?/
>
>     /dbcassa_base.cpp: In function ?ColumnVecPtr
>     cassa_translate_query(const db1_con_t*, str* const*, const
>     db_val_t*, str* const*, int, int, int*)?:/
>
>     /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of
>     type ?int?, but argument 7 has type
>     ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka
>     long unsigned int}? [-Wformat]/
>
>     /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of
>     type ?int?, but argument 5 has type
>     ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka
>     long unsigned int}? [-Wformat]/
>
>     /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of
>     type ?int?, but argument 6 has type
>     ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka
>     long unsigned int}? [-Wformat]/
>
>     /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of
>     type ?int?, but argument 4 has type
>     ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka
>     long unsigned int}? [-Wformat]/
>
>     /dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of
>     type ?int?, but argument 4 has type
>     ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka
>     long unsigned int}? [-Wformat]/
>
>     /../../lib/srdb1/db_res.h: At global scope:/
>
>     /../../lib/srdb1/db_res.h:105:19: warning: inline function
>     ?db1_res_t* db_new_result()? used but never defined [enabled by
>     default]/
>
>     /../../lib/srdb1/db_res.h:121:12: warning: inline function ?int
>     db_allocate_columns(db1_res_t*, unsigned int)? used but never
>     defined [enabled by default]/
>
>     /../../lib/srdb1/db_res.h:98:12: warning: inline function ?int
>     db_free_columns(db1_res_t*)? used but never defined [enabled by
>     default]/
>
>     /../../lib/srdb1/db_res.h:129:12: warning: inline function ?int
>     db_allocate_rows(db1_res_t*)? used but never defined [enabled by
>     default]/
>
>     /../../lib/srdb1/db_row.h:74:12: warning: inline function ?int
>     db_allocate_row(const db1_res_t*, db_row_t*)? used but never
>     defined [enabled by default]/
>
>     /../../lib/srdb1/db_res.h:111:12: warning: inline function ?int
>     db_free_result(db1_res_t*)? used but never defined [enabled by
>     default]/
>
>     /make[1]: *** [dbcassa_base.o] Error 1/
>
>     /make: *** [install-modules] Error 1/
>
>     //
>
>     //
>
>     Anybody has got any ideas?
>
>     Thanks in advance,
>
>     Bou
>


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From jcastro at instant.com.br  Tue Aug 14 16:40:04 2012
From: jcastro at instant.com.br (Juan Castro)
Date: Tue, 14 Aug 2012 11:40:04 -0300
Subject: [SR-Users] A much more modest initial goal: get Kamailio dev
	version up and running
Message-ID: <CALRpQ=64rs6fBc26jz4gtFsHyUxO4GXPcuJCWxgUpesW1P0BsA@mail.gmail.com>

...with mysql user auth. I created the DB with the script in
scripts/mysql in the sources, but apparently it doesn't create a
"version" table. The startup complains about that.

Where is the schema for that table, and how should have done things instead?

mysql> show tables;
+-----------------------+
| Tables_in_ser         |
+-----------------------+
| acc                   |
| attr_types            |
| contact_attrs         |
| cpl                   |
| credentials           |
| customers             |
| domain                |
| domain_attrs          |
| domain_settings       |
| global_attrs          |
| grp                   |
| gw                    |
| gw_grp                |
| i18n                  |
| ipmatch               |
| lcr                   |
| location              |
| missed_calls          |
| offline_winfo         |
| pdt                   |
| phonebook             |
| presentity            |
| presentity_contact    |
| presentity_extensions |
| presentity_notes      |
| rls_subscription      |
| rls_vs                |
| rls_vs_names          |
| sd_attrs              |
| silo                  |
| speed_dial            |
| trusted               |
| tuple_extensions      |
| tuple_notes           |
| uri                   |
| uri_attrs             |
| user_attrs            |
| watcherinfo           |
+-----------------------+
38 rows in set (0.00 sec)


-- 
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regi?es: (11)4063-6100


From oleksandr.shchirskyi at globallogic.com  Tue Aug 14 17:10:13 2012
From: oleksandr.shchirskyi at globallogic.com (Oleksandr Shchirskyi)
Date: Tue, 14 Aug 2012 18:10:13 +0300
Subject: [SR-Users] sms module & IMS
Message-ID: <CAGrTonSCfYEmMPqvLvwx9GWQ=eKE1WADbxfO1D0mhZaScoBrLQ@mail.gmail.com>

Hi,

I'm using OpenIMSCore based on the SER. At the moment I try to add
SMS-GW to it.
I've found, that SER has sms module, and I tried to use it. But,
unfortunately, I couldn't make it work.

It's not very clear for me how SIP MESSAGEs can be routed to the sms
module. This module works not as application server (or I'm wrong?)

At the moment I have such configuration in the scscf.cfg:

==============================
modparam("sms","modems","Nokia [d=/dev/ttyACM0;b=9600]")
modparam("sms","networks","D1[m=10]")
modparam("sms","links","Nokia[D1]")
modparam("sms","domain","open-ims.test")
modparam("sms","max_sms_parts",3)
modparam("sms","use_contact",1)

# main routing logic

route{
        if(method=="MESSAGE"){
                sms_send_msg_to_net("D1");
        }
...
}

==============================

As a hardware GSM gateway my cell phone is using.
As I see, nothing happens when s-cscf processes SIP MESSAGEs. I don't
see that these messages are forwarded to the sms module, or smth like
that.

Final question - is it possible to use this module as IP-SMS-GW for
the OpenIMSCore? And how?

I would be very appreciated for any help.
Thank you in advance.

-- 
Oleksandr Shchirskyi | Software Testing Engineer
GlobalLogic Inc. | Innovation by Design
ARGENTINA | CHILE | CHINA | INDIA | ISRAEL | UKRAINE | UK | USA
Office: +044.492.96.95, (ext. 4296)
www.globallogic.com


From jcastro at instant.com.br  Tue Aug 14 17:16:15 2012
From: jcastro at instant.com.br (Juan Castro)
Date: Tue, 14 Aug 2012 12:16:15 -0300
Subject: [SR-Users] A much more modest initial goal: get Kamailio dev
	version up and running
In-Reply-To: <CALRpQ=64rs6fBc26jz4gtFsHyUxO4GXPcuJCWxgUpesW1P0BsA@mail.gmail.com>
References: <CALRpQ=64rs6fBc26jz4gtFsHyUxO4GXPcuJCWxgUpesW1P0BsA@mail.gmail.com>
Message-ID: <CALRpQ=7T5KgVc91cRDtSa=d11fsm0fAjF4SiX-Wm4=S_rswctQ@mail.gmail.com>

Never mind, I found it out, what I had to do is create the database
with "kamdbctl init". Please don't beat me up.

On Tue, Aug 14, 2012 at 11:40 AM, Juan Castro <jcastro at instant.com.br> wrote:
> ...with mysql user auth. I created the DB with the script in
> scripts/mysql in the sources, but apparently it doesn't create a
> "version" table. The startup complains about that.
>
> Where is the schema for that table, and how should have done things instead?


From brandon at cryy.com  Tue Aug 14 19:16:51 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 10:16:51 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <4EEF96B4.2050806@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
Message-ID: <CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>

Sorry to wake up an old thread.

However - I am looking to export this $timef function to the param
initialization for accounting, i.e. db_table_acc

modparam("db_table_acc", "acc_$ftime(Ymd)");

I've looked into completing this myself however I simply am not familiar
enough at this point between the three different modules that it would take
to implement this (acc, dbsr1, pv).

Look forward to any help / insight you may be able to provide.

Thanks as always!

Sincerely,
Brandon Armstead

On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

> Hello,
>
>
> On 12/19/11 7:50 PM, Andreas Granig wrote:
>
>> Hi Daniel,
>>
>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>
>>> I don't know what are all the functions you think of, but for the
>>> example provided above, config file does it easy right now. There is a
>>> pseudo-variable that gives broken-time attribute that can be used with
>>> avp_check(), iirc, should be:
>>>
>>> avp_db_load(...);
>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>
>>> Of course there is the option of doing while loop, but maybe gets to
>>> large for desired config file.
>>>
>> This is pretty much what I had in mind with my new functions/module, but
>> I've completely overlooked that PV when searching the docs for this
>> feature. Thank you very much for pointing that out!
>>
> for sake of public knowledge, just to add on time specific features: there
> is also $timef(format) which returns current time attributes based on
> strftime specifiers -- its documentation was missing, I just added it.
> Also, there is a transformation {s.ftime,format) which can take any integer
> variable holding timestamp and return value based on strftime format.
>
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
> ______________________________**_________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>
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From miconda at gmail.com  Tue Aug 14 19:27:10 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 19:27:10 +0200
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
Message-ID: <502A8A6E.906@gmail.com>

Hello,

this functionality should be already there. Doesn't work for you?

Cheers,
Daniel

On 8/14/12 7:16 PM, Brandon Armstead wrote:
> Sorry to wake up an old thread.
>
> However - I am looking to export this $timef function to the param 
> initialization for accounting, i.e. db_table_acc
>
> modparam("db_table_acc", "acc_$ftime(Ymd)");
>
> I've looked into completing this myself however I simply am not 
> familiar enough at this point between the three different modules that 
> it would take to implement this (acc, dbsr1, pv).
>
> Look forward to any help / insight you may be able to provide.
>
> Thanks as always!
>
> Sincerely,
> Brandon Armstead
>
> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>
>     On 12/19/11 7:50 PM, Andreas Granig wrote:
>
>         Hi Daniel,
>
>         On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>
>             I don't know what are all the functions you think of, but
>             for the
>             example provided above, config file does it easy right
>             now. There is a
>             pseudo-variable that gives broken-time attribute that can
>             be used with
>             avp_check(), iirc, should be:
>
>             avp_db_load(...);
>             if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) {
>             do CF }
>
>             Of course there is the option of doing while loop, but
>             maybe gets to
>             large for desired config file.
>
>         This is pretty much what I had in mind with my new
>         functions/module, but
>         I've completely overlooked that PV when searching the docs for
>         this
>         feature. Thank you very much for pointing that out!
>
>     for sake of public knowledge, just to add on time specific
>     features: there is also $timef(format) which returns current time
>     attributes based on strftime specifiers -- its documentation was
>     missing, I just added it. Also, there is a transformation
>     {s.ftime,format) which can take any integer variable holding
>     timestamp and return value based on strftime format.
>
>
>     Cheers,
>     Daniel
>
>     -- 
>     Daniel-Constantin Mierla -- http://www.asipto.com
>     http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
>     _______________________________________________
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>     list
>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From brandon at cryy.com  Tue Aug 14 19:30:28 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 10:30:28 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502A8A6E.906@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
Message-ID: <5D17F9F5-8B9F-409A-965A-22994A0AB77F@cryy.com>

Daniel,

   It ONLY works with the $time avp

Sent from my iPhone

On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hello,
> 
> this functionality should be already there. Doesn't work for you?
> 
> Cheers,
> Daniel
> 
> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>> Sorry to wake up an old thread.
>> 
>> However - I am looking to export this $timef function to the param initialization for accounting, i.e. db_table_acc
>> 
>> modparam("db_table_acc", "acc_$ftime(Ymd)");
>> 
>> I've looked into completing this myself however I simply am not familiar enough at this point between the three different modules that it would take to implement this (acc, dbsr1, pv).
>> 
>> Look forward to any help / insight you may be able to provide.
>> 
>> Thanks as always!
>> 
>> Sincerely,
>> Brandon Armstead
>> 
>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>> Hello,
>> 
>> 
>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>> Hi Daniel,
>> 
>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>> I don't know what are all the functions you think of, but for the
>> example provided above, config file does it easy right now. There is a
>> pseudo-variable that gives broken-time attribute that can be used with
>> avp_check(), iirc, should be:
>> 
>> avp_db_load(...);
>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>> 
>> Of course there is the option of doing while loop, but maybe gets to
>> large for desired config file.
>> This is pretty much what I had in mind with my new functions/module, but
>> I've completely overlooked that PV when searching the docs for this
>> feature. Thank you very much for pointing that out!
>> for sake of public knowledge, just to add on time specific features: there is also $timef(format) which returns current time attributes based on strftime specifiers -- its documentation was missing, I just added it. Also, there is a transformation {s.ftime,format) which can take any integer variable holding timestamp and return value based on strftime format.
>> 
>> 
>> Cheers,
>> Daniel
>> 
>> -- 
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>> 
>> 
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> 
> 
> -- 
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
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From brandon at cryy.com  Tue Aug 14 19:34:35 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 10:34:35 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502A8A6E.906@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
Message-ID: <69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>

Let me also add that I am using 3.2 but I see the commit was quite some time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!

Sent from my iPhone

On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hello,
> 
> this functionality should be already there. Doesn't work for you?
> 
> Cheers,
> Daniel
> 
> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>> Sorry to wake up an old thread.
>> 
>> However - I am looking to export this $timef function to the param initialization for accounting, i.e. db_table_acc
>> 
>> modparam("db_table_acc", "acc_$ftime(Ymd)");
>> 
>> I've looked into completing this myself however I simply am not familiar enough at this point between the three different modules that it would take to implement this (acc, dbsr1, pv).
>> 
>> Look forward to any help / insight you may be able to provide.
>> 
>> Thanks as always!
>> 
>> Sincerely,
>> Brandon Armstead
>> 
>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>> Hello,
>> 
>> 
>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>> Hi Daniel,
>> 
>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>> I don't know what are all the functions you think of, but for the
>> example provided above, config file does it easy right now. There is a
>> pseudo-variable that gives broken-time attribute that can be used with
>> avp_check(), iirc, should be:
>> 
>> avp_db_load(...);
>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>> 
>> Of course there is the option of doing while loop, but maybe gets to
>> large for desired config file.
>> This is pretty much what I had in mind with my new functions/module, but
>> I've completely overlooked that PV when searching the docs for this
>> feature. Thank you very much for pointing that out!
>> for sake of public knowledge, just to add on time specific features: there is also $timef(format) which returns current time attributes based on strftime specifiers -- its documentation was missing, I just added it. Also, there is a transformation {s.ftime,format) which can take any integer variable holding timestamp and return value based on strftime format.
>> 
>> 
>> Cheers,
>> Daniel
>> 
>> -- 
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>> 
>> 
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> 
> 
> -- 
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
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From miconda at gmail.com  Tue Aug 14 21:04:31 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 21:04:31 +0200
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
Message-ID: <502AA13F.4030203@gmail.com>

Hello,

checked the sources and it seems that Juha reverted this feature with 
the commit:

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca

I guess it was accidentally, I will ask on devels list.

Cheers,
Daniel


On 8/14/12 7:34 PM, Brandon Armstead wrote:
> Let me also add that I am using 3.2 but I see the commit was quite 
> some time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>
> Sent from my iPhone
>
> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>> Hello,
>>
>> this functionality should be already there. Doesn't work for you?
>>
>> Cheers,
>> Daniel
>>
>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>> Sorry to wake up an old thread.
>>>
>>> However - I am looking to export this $timef function to the param 
>>> initialization for accounting, i.e. db_table_acc
>>>
>>> modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>
>>> I've looked into completing this myself however I simply am not 
>>> familiar enough at this point between the three different modules 
>>> that it would take to implement this (acc, dbsr1, pv).
>>>
>>> Look forward to any help / insight you may be able to provide.
>>>
>>> Thanks as always!
>>>
>>> Sincerely,
>>> Brandon Armstead
>>>
>>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla 
>>> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>
>>>     Hello,
>>>
>>>
>>>     On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>
>>>         Hi Daniel,
>>>
>>>         On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>
>>>             I don't know what are all the functions you think of,
>>>             but for the
>>>             example provided above, config file does it easy right
>>>             now. There is a
>>>             pseudo-variable that gives broken-time attribute that
>>>             can be used with
>>>             avp_check(), iirc, should be:
>>>
>>>             avp_db_load(...);
>>>             if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g"))
>>>             { do CF }
>>>
>>>             Of course there is the option of doing while loop, but
>>>             maybe gets to
>>>             large for desired config file.
>>>
>>>         This is pretty much what I had in mind with my new
>>>         functions/module, but
>>>         I've completely overlooked that PV when searching the docs
>>>         for this
>>>         feature. Thank you very much for pointing that out!
>>>
>>>     for sake of public knowledge, just to add on time specific
>>>     features: there is also $timef(format) which returns current
>>>     time attributes based on strftime specifiers -- its
>>>     documentation was missing, I just added it. Also, there is a
>>>     transformation {s.ftime,format) which can take any integer
>>>     variable holding timestamp and return value based on strftime
>>>     format.
>>>
>>>
>>>     Cheers,
>>>     Daniel
>>>
>>>     -- 
>>>     Daniel-Constantin Mierla -- http://www.asipto.com
>>>     http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>
>>>
>>>     _______________________________________________
>>>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>>     mailing list
>>>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>>>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> -- 
>> Daniel-Constantin Mierla -http://www.asipto.com
>> http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From brandon at cryy.com  Tue Aug 14 21:21:52 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 12:21:52 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502AA13F.4030203@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
Message-ID: <CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>

Daniel,

   In my research I saw that commit as well but figured it was some kind of
weird merging error.  Thanks for your time!  Look forward to hearing back
from you guys.

Sincerely,
Brandon Armstead

On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

>  Hello,
>
> checked the sources and it seems that Juha reverted this feature with the
> commit:
>
>
> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>
> I guess it was accidentally, I will ask on devels list.
>
> Cheers,
> Daniel
>
>
>
> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>
> Let me also add that I am using 3.2 but I see the commit was quite some
> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>
> Sent from my iPhone
>
> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <miconda at gmail.com>
> wrote:
>
>   Hello,
>
> this functionality should be already there. Doesn't work for you?
>
> Cheers,
> Daniel
>
> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>
> Sorry to wake up an old thread.
>
>  However - I am looking to export this $timef function to the param
> initialization for accounting, i.e. db_table_acc
>
>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>
>  I've looked into completing this myself however I simply am not familiar
> enough at this point between the three different modules that it would take
> to implement this (acc, dbsr1, pv).
>
>  Look forward to any help / insight you may be able to provide.
>
>  Thanks as always!
>
>  Sincerely,
> Brandon Armstead
>
> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>> Hello,
>>
>>
>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>
>>> Hi Daniel,
>>>
>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>
>>>> I don't know what are all the functions you think of, but for the
>>>> example provided above, config file does it easy right now. There is a
>>>> pseudo-variable that gives broken-time attribute that can be used with
>>>> avp_check(), iirc, should be:
>>>>
>>>> avp_db_load(...);
>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>
>>>> Of course there is the option of doing while loop, but maybe gets to
>>>> large for desired config file.
>>>>
>>> This is pretty much what I had in mind with my new functions/module, but
>>> I've completely overlooked that PV when searching the docs for this
>>> feature. Thank you very much for pointing that out!
>>>
>>  for sake of public knowledge, just to add on time specific features:
>> there is also $timef(format) which returns current time attributes based on
>> strftime specifiers -- its documentation was missing, I just added it.
>> Also, there is a transformation {s.ftime,format) which can take any integer
>> variable holding timestamp and return value based on strftime format.
>>
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From miconda at gmail.com  Tue Aug 14 21:48:01 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 21:48:01 +0200
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
Message-ID: <502AAB71.5060508@gmail.com>

Hello,

I reapplied the patch (cherry-picked from the initial one) to the master 
branch.

Can you test that and see if it works fine?

You can cherry-picked to your branch, try:

git pull origin
git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91

If it applies fine, compiles and the test go ok, then I will backport as 
soon as Juha confirms that was no solid reason in discarding this 
feature by his commit.

Cheers,
Daniel

On 8/14/12 9:21 PM, Brandon Armstead wrote:
> Daniel,
>
>    In my research I saw that commit as well but figured it was some 
> kind of weird merging error.  Thanks for your time!  Look forward to 
> hearing back from you guys.
>
> Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     checked the sources and it seems that Juha reverted this feature
>     with the commit:
>
>     http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>
>     I guess it was accidentally, I will ask on devels list.
>
>     Cheers,
>     Daniel
>
>
>
>     On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>     Let me also add that I am using 3.2 but I see the commit was
>>     quite some time ago so I have a feeling it wouldn't work in 3.3
>>     either.  Thanks!
>>
>>     Sent from my iPhone
>>
>>     On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>>     Hello,
>>>
>>>     this functionality should be already there. Doesn't work for you?
>>>
>>>     Cheers,
>>>     Daniel
>>>
>>>     On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>     Sorry to wake up an old thread.
>>>>
>>>>     However - I am looking to export this $timef function to the
>>>>     param initialization for accounting, i.e. db_table_acc
>>>>
>>>>     modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>
>>>>     I've looked into completing this myself however I simply am not
>>>>     familiar enough at this point between the three different
>>>>     modules that it would take to implement this (acc, dbsr1, pv).
>>>>
>>>>     Look forward to any help / insight you may be able to provide.
>>>>
>>>>     Thanks as always!
>>>>
>>>>     Sincerely,
>>>>     Brandon Armstead
>>>>
>>>>     On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla
>>>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>>
>>>>         Hello,
>>>>
>>>>
>>>>         On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>
>>>>             Hi Daniel,
>>>>
>>>>             On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>
>>>>                 I don't know what are all the functions you think
>>>>                 of, but for the
>>>>                 example provided above, config file does it easy
>>>>                 right now. There is a
>>>>                 pseudo-variable that gives broken-time attribute
>>>>                 that can be used with
>>>>                 avp_check(), iirc, should be:
>>>>
>>>>                 avp_db_load(...);
>>>>                 if(avp_check("$time(wday)",
>>>>                 "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>
>>>>                 Of course there is the option of doing while loop,
>>>>                 but maybe gets to
>>>>                 large for desired config file.
>>>>
>>>>             This is pretty much what I had in mind with my new
>>>>             functions/module, but
>>>>             I've completely overlooked that PV when searching the
>>>>             docs for this
>>>>             feature. Thank you very much for pointing that out!
>>>>
>>>>         for sake of public knowledge, just to add on time specific
>>>>         features: there is also $timef(format) which returns
>>>>         current time attributes based on strftime specifiers -- its
>>>>         documentation was missing, I just added it. Also, there is
>>>>         a transformation {s.ftime,format) which can take any
>>>>         integer variable holding timestamp and return value based
>>>>         on strftime format.
>>>>
>>>>
>>>>         Cheers,
>>>>         Daniel
>>>>
>>>>         -- 
>>>>         Daniel-Constantin Mierla -- http://www.asipto.com
>>>>         http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>>
>>>>
>>>>         _______________________________________________
>>>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>>>         mailing list
>>>>         sr-users at lists.sip-router.org
>>>>         <mailto:sr-users at lists.sip-router.org>
>>>>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>>     -- 
>>>     Daniel-Constantin Mierla -http://www.asipto.com
>>>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>     Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>     Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>     Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From brandon at cryy.com  Tue Aug 14 21:55:20 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 12:55:20 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502AAB71.5060508@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
Message-ID: <CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>

Daniel,

   Patch applies fine - it still does not seem to take when using $timef,
i..e

modparam("acc", "db_table_acc", "acc_$timef(Ymd)")

It looks as if the $timef is not being interpreted and is simply writing to
acc'ing i.e. INSERT INTO acc_$timef

Thanks!

Sincerely,
Brandon Armstead
On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

>  Hello,
>
> I reapplied the patch (cherry-picked from the initial one) to the master
> branch.
>
> Can you test that and see if it works fine?
>
> You can cherry-picked to your branch, try:
>
> git pull origin
> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>
> If it applies fine, compiles and the test go ok, then I will backport as
> soon as Juha confirms that was no solid reason in discarding this feature
> by his commit.
>
> Cheers,
> Daniel
>
>
> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>
> Daniel,
>
>     In my research I saw that commit as well but figured it was some kind
> of weird merging error.  Thanks for your time!  Look forward to hearing
> back from you guys.
>
>  Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> checked the sources and it seems that Juha reverted this feature with the
>> commit:
>>
>>
>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>
>> I guess it was accidentally, I will ask on devels list.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>
>> Let me also add that I am using 3.2 but I see the commit was quite some
>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>
>> Sent from my iPhone
>>
>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <miconda at gmail.com>
>> wrote:
>>
>>   Hello,
>>
>> this functionality should be already there. Doesn't work for you?
>>
>> Cheers,
>> Daniel
>>
>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>
>> Sorry to wake up an old thread.
>>
>>  However - I am looking to export this $timef function to the param
>> initialization for accounting, i.e. db_table_acc
>>
>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>
>>  I've looked into completing this myself however I simply am not
>> familiar enough at this point between the three different modules that it
>> would take to implement this (acc, dbsr1, pv).
>>
>>  Look forward to any help / insight you may be able to provide.
>>
>>  Thanks as always!
>>
>>  Sincerely,
>> Brandon Armstead
>>
>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>
>>>> Hi Daniel,
>>>>
>>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>
>>>>> I don't know what are all the functions you think of, but for the
>>>>> example provided above, config file does it easy right now. There is a
>>>>> pseudo-variable that gives broken-time attribute that can be used with
>>>>> avp_check(), iirc, should be:
>>>>>
>>>>> avp_db_load(...);
>>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>
>>>>> Of course there is the option of doing while loop, but maybe gets to
>>>>> large for desired config file.
>>>>>
>>>> This is pretty much what I had in mind with my new functions/module, but
>>>> I've completely overlooked that PV when searching the docs for this
>>>> feature. Thank you very much for pointing that out!
>>>>
>>>  for sake of public knowledge, just to add on time specific features:
>>> there is also $timef(format) which returns current time attributes based on
>>> strftime specifiers -- its documentation was missing, I just added it.
>>> Also, there is a transformation {s.ftime,format) which can take any integer
>>> variable holding timestamp and return value based on strftime format.
>>>
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From miconda at gmail.com  Tue Aug 14 22:11:17 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 22:11:17 +0200
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
	<CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
Message-ID: <502AB0E5.10303@gmail.com>

Hello,

are you sure you re-installed and the right acc.so is used?

The support is for generic PV, if it works with one it should work with 
any. Which one does work for you?

Can you give exact table name as printed in the SQL query? Does it have 
the parenthesis and the format string?

Cheers,
Daniel

On 8/14/12 9:55 PM, Brandon Armstead wrote:
> Daniel,
>
>    Patch applies fine - it still does not seem to take when using 
> $timef, i..e
>
> modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>
> It looks as if the $timef is not being interpreted and is simply 
> writing to acc'ing i.e. INSERT INTO acc_$timef
>
> Thanks!
>
> Sincerely,
> Brandon Armstead
> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     I reapplied the patch (cherry-picked from the initial one) to the
>     master branch.
>
>     Can you test that and see if it works fine?
>
>     You can cherry-picked to your branch, try:
>
>     git pull origin
>     git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>
>     If it applies fine, compiles and the test go ok, then I will
>     backport as soon as Juha confirms that was no solid reason in
>     discarding this feature by his commit.
>
>     Cheers,
>     Daniel
>
>
>     On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>     Daniel,
>>
>>        In my research I saw that commit as well but figured it was
>>     some kind of weird merging error.  Thanks for your time!  Look
>>     forward to hearing back from you guys.
>>
>>     Sincerely,
>>     Brandon Armstead
>>
>>     On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         checked the sources and it seems that Juha reverted this
>>         feature with the commit:
>>
>>         http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>
>>         I guess it was accidentally, I will ask on devels list.
>>
>>         Cheers,
>>         Daniel
>>
>>
>>
>>         On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>         Let me also add that I am using 3.2 but I see the commit was
>>>         quite some time ago so I have a feeling it wouldn't work in
>>>         3.3 either.  Thanks!
>>>
>>>         Sent from my iPhone
>>>
>>>         On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla
>>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>
>>>>         Hello,
>>>>
>>>>         this functionality should be already there. Doesn't work
>>>>         for you?
>>>>
>>>>         Cheers,
>>>>         Daniel
>>>>
>>>>         On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>>         Sorry to wake up an old thread.
>>>>>
>>>>>         However - I am looking to export this $timef function to
>>>>>         the param initialization for accounting, i.e. db_table_acc
>>>>>
>>>>>         modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>>
>>>>>         I've looked into completing this myself however I simply
>>>>>         am not familiar enough at this point between the three
>>>>>         different modules that it would take to implement this
>>>>>         (acc, dbsr1, pv).
>>>>>
>>>>>         Look forward to any help / insight you may be able to provide.
>>>>>
>>>>>         Thanks as always!
>>>>>
>>>>>         Sincerely,
>>>>>         Brandon Armstead
>>>>>
>>>>>         On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla
>>>>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>>>
>>>>>             Hello,
>>>>>
>>>>>
>>>>>             On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>>
>>>>>                 Hi Daniel,
>>>>>
>>>>>                 On 12/19/2011 07:29 PM, Daniel-Constantin Mierla
>>>>>                 wrote:
>>>>>
>>>>>                     I don't know what are all the functions you
>>>>>                     think of, but for the
>>>>>                     example provided above, config file does it
>>>>>                     easy right now. There is a
>>>>>                     pseudo-variable that gives broken-time
>>>>>                     attribute that can be used with
>>>>>                     avp_check(), iirc, should be:
>>>>>
>>>>>                     avp_db_load(...);
>>>>>                     if(avp_check("$time(wday)",
>>>>>                     "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>
>>>>>                     Of course there is the option of doing while
>>>>>                     loop, but maybe gets to
>>>>>                     large for desired config file.
>>>>>
>>>>>                 This is pretty much what I had in mind with my new
>>>>>                 functions/module, but
>>>>>                 I've completely overlooked that PV when searching
>>>>>                 the docs for this
>>>>>                 feature. Thank you very much for pointing that out!
>>>>>
>>>>>             for sake of public knowledge, just to add on time
>>>>>             specific features: there is also $timef(format) which
>>>>>             returns current time attributes based on strftime
>>>>>             specifiers -- its documentation was missing, I just
>>>>>             added it. Also, there is a transformation
>>>>>             {s.ftime,format) which can take any integer variable
>>>>>             holding timestamp and return value based on strftime
>>>>>             format.
>>>>>
>>>>>
>>>>>             Cheers,
>>>>>             Daniel
>>>>>
>>>>>             -- 
>>>>>             Daniel-Constantin Mierla -- http://www.asipto.com
>>>>>             http://linkedin.com/in/miconda --
>>>>>             http://twitter.com/miconda
>>>>>
>>>>>
>>>>>             _______________________________________________
>>>>>             SIP Express Router (SER) and Kamailio (OpenSER) -
>>>>>             sr-users mailing list
>>>>>             sr-users at lists.sip-router.org
>>>>>             <mailto:sr-users at lists.sip-router.org>
>>>>>             http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>
>>>>         -- 
>>>>         Daniel-Constantin Mierla -http://www.asipto.com
>>>>         http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>>         Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>>         Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>
>>         -- 
>>         Daniel-Constantin Mierla -http://www.asipto.com
>>         http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>         Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>         Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>     Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From brandon at cryy.com  Tue Aug 14 22:17:23 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 13:17:23 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502AB0E5.10303@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
	<CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
	<502AB0E5.10303@gmail.com>
Message-ID: <CACr1DbDt2N5SAYLZF40GF7FAHDAzc_yLVbCP+f=SvEdWm1955w@mail.gmail.com>

Daniel,

    My apologies - forgot to make clean.  However still no cigar.

Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' doesn't
exist
Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc [acc.c:405]:
failed to insert into database

Sincerely,
Brandon Armstead

On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> are you sure you re-installed and the right acc.so is used?
>
> The support is for generic PV, if it works with one it should work with
> any. Which one does work for you?
>
> Can you give exact table name as printed in the SQL query? Does it have
> the parenthesis and the format string?
>
> Cheers,
> Daniel
>
>
> On 8/14/12 9:55 PM, Brandon Armstead wrote:
>
> Daniel,
>
>     Patch applies fine - it still does not seem to take when using
> $timef, i..e
>
>  modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>
>  It looks as if the $timef is not being interpreted and is simply writing
> to acc'ing i.e. INSERT INTO acc_$timef
>
>  Thanks!
>
>  Sincerely,
> Brandon Armstead
> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> I reapplied the patch (cherry-picked from the initial one) to the master
>> branch.
>>
>> Can you test that and see if it works fine?
>>
>> You can cherry-picked to your branch, try:
>>
>> git pull origin
>> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>
>> If it applies fine, compiles and the test go ok, then I will backport as
>> soon as Juha confirms that was no solid reason in discarding this feature
>> by his commit.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>
>> Daniel,
>>
>>     In my research I saw that commit as well but figured it was some
>> kind of weird merging error.  Thanks for your time!  Look forward to
>> hearing back from you guys.
>>
>>  Sincerely,
>> Brandon Armstead
>>
>> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> checked the sources and it seems that Juha reverted this feature with
>>> the commit:
>>>
>>>
>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>
>>> I guess it was accidentally, I will ask on devels list.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>
>>> Let me also add that I am using 3.2 but I see the commit was quite some
>>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>>
>>> Sent from my iPhone
>>>
>>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <
>>> miconda at gmail.com> wrote:
>>>
>>>   Hello,
>>>
>>> this functionality should be already there. Doesn't work for you?
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>
>>> Sorry to wake up an old thread.
>>>
>>>  However - I am looking to export this $timef function to the param
>>> initialization for accounting, i.e. db_table_acc
>>>
>>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>
>>>  I've looked into completing this myself however I simply am not
>>> familiar enough at this point between the three different modules that it
>>> would take to implement this (acc, dbsr1, pv).
>>>
>>>  Look forward to any help / insight you may be able to provide.
>>>
>>>  Thanks as always!
>>>
>>>  Sincerely,
>>> Brandon Armstead
>>>
>>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
>>> miconda at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>>
>>>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>
>>>>> Hi Daniel,
>>>>>
>>>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>>
>>>>>> I don't know what are all the functions you think of, but for the
>>>>>> example provided above, config file does it easy right now. There is a
>>>>>> pseudo-variable that gives broken-time attribute that can be used with
>>>>>> avp_check(), iirc, should be:
>>>>>>
>>>>>> avp_db_load(...);
>>>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>>
>>>>>> Of course there is the option of doing while loop, but maybe gets to
>>>>>> large for desired config file.
>>>>>>
>>>>> This is pretty much what I had in mind with my new functions/module,
>>>>> but
>>>>> I've completely overlooked that PV when searching the docs for this
>>>>> feature. Thank you very much for pointing that out!
>>>>>
>>>>  for sake of public knowledge, just to add on time specific features:
>>>> there is also $timef(format) which returns current time attributes based on
>>>> strftime specifiers -- its documentation was missing, I just added it.
>>>> Also, there is a transformation {s.ftime,format) which can take any integer
>>>> variable holding timestamp and return value based on strftime format.
>>>>
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> --
>>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From miconda at gmail.com  Tue Aug 14 22:22:05 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 14 Aug 2012 22:22:05 +0200
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <CACr1DbDt2N5SAYLZF40GF7FAHDAzc_yLVbCP+f=SvEdWm1955w@mail.gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
	<CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
	<502AB0E5.10303@gmail.com>
	<CACr1DbDt2N5SAYLZF40GF7FAHDAzc_yLVbCP+f=SvEdWm1955w@mail.gmail.com>
Message-ID: <502AB36D.7080006@gmail.com>

Hello,

the $timef(...) is evaluated, but you don't have time specifiers there, 
only static letters. You have to use % in front of the letters, as I 
guess from the example:

http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables

Try acc_$timef(%Y%m%d).

Cheers,
Daniel


On 8/14/12 10:17 PM, Brandon Armstead wrote:
> Daniel,
>
>     My apologies - forgot to make clean.  However still no cigar.
>
> Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql 
> [km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' 
> doesn't exist
> Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc 
> [acc.c:405]: failed to insert into database
>
> Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     are you sure you re-installed and the right acc.so is used?
>
>     The support is for generic PV, if it works with one it should work
>     with any. Which one does work for you?
>
>     Can you give exact table name as printed in the SQL query? Does it
>     have the parenthesis and the format string?
>
>     Cheers,
>     Daniel
>
>
>     On 8/14/12 9:55 PM, Brandon Armstead wrote:
>>     Daniel,
>>
>>        Patch applies fine - it still does not seem to take when using
>>     $timef, i..e
>>
>>     modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>>
>>     It looks as if the $timef is not being interpreted and is simply
>>     writing to acc'ing i.e. INSERT INTO acc_$timef
>>
>>     Thanks!
>>
>>     Sincerely,
>>     Brandon Armstead
>>     On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         I reapplied the patch (cherry-picked from the initial one) to
>>         the master branch.
>>
>>         Can you test that and see if it works fine?
>>
>>         You can cherry-picked to your branch, try:
>>
>>         git pull origin
>>         git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>
>>         If it applies fine, compiles and the test go ok, then I will
>>         backport as soon as Juha confirms that was no solid reason in
>>         discarding this feature by his commit.
>>
>>         Cheers,
>>         Daniel
>>
>>
>>         On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>>         Daniel,
>>>
>>>            In my research I saw that commit as well but figured it
>>>         was some kind of weird merging error.  Thanks for your time!
>>>          Look forward to hearing back from you guys.
>>>
>>>         Sincerely,
>>>         Brandon Armstead
>>>
>>>         On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla
>>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>
>>>             Hello,
>>>
>>>             checked the sources and it seems that Juha reverted this
>>>             feature with the commit:
>>>
>>>             http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>
>>>             I guess it was accidentally, I will ask on devels list.
>>>
>>>             Cheers,
>>>             Daniel
>>>
>>>
>>>
>>>             On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>>             Let me also add that I am using 3.2 but I see the
>>>>             commit was quite some time ago so I have a feeling it
>>>>             wouldn't work in 3.3 either.  Thanks!
>>>>
>>>>             Sent from my iPhone
>>>>
>>>>             On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla
>>>>             <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>>
>>>>>             Hello,
>>>>>
>>>>>             this functionality should be already there. Doesn't
>>>>>             work for you?
>>>>>
>>>>>             Cheers,
>>>>>             Daniel
>>>>>
>>>>>             On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>>>             Sorry to wake up an old thread.
>>>>>>
>>>>>>             However - I am looking to export this $timef function
>>>>>>             to the param initialization for accounting, i.e.
>>>>>>             db_table_acc
>>>>>>
>>>>>>             modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>>>
>>>>>>             I've looked into completing this myself however I
>>>>>>             simply am not familiar enough at this point between
>>>>>>             the three different modules that it would take to
>>>>>>             implement this (acc, dbsr1, pv).
>>>>>>
>>>>>>             Look forward to any help / insight you may be able to
>>>>>>             provide.
>>>>>>
>>>>>>             Thanks as always!
>>>>>>
>>>>>>             Sincerely,
>>>>>>             Brandon Armstead
>>>>>>
>>>>>>             On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin
>>>>>>             Mierla <miconda at gmail.com <mailto:miconda at gmail.com>>
>>>>>>             wrote:
>>>>>>
>>>>>>                 Hello,
>>>>>>
>>>>>>
>>>>>>                 On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>>>
>>>>>>                     Hi Daniel,
>>>>>>
>>>>>>                     On 12/19/2011 07:29 PM, Daniel-Constantin
>>>>>>                     Mierla wrote:
>>>>>>
>>>>>>                         I don't know what are all the functions
>>>>>>                         you think of, but for the
>>>>>>                         example provided above, config file does
>>>>>>                         it easy right now. There is a
>>>>>>                         pseudo-variable that gives broken-time
>>>>>>                         attribute that can be used with
>>>>>>                         avp_check(), iirc, should be:
>>>>>>
>>>>>>                         avp_db_load(...);
>>>>>>                         if(avp_check("$time(wday)",
>>>>>>                         "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>>
>>>>>>                         Of course there is the option of doing
>>>>>>                         while loop, but maybe gets to
>>>>>>                         large for desired config file.
>>>>>>
>>>>>>                     This is pretty much what I had in mind with
>>>>>>                     my new functions/module, but
>>>>>>                     I've completely overlooked that PV when
>>>>>>                     searching the docs for this
>>>>>>                     feature. Thank you very much for pointing
>>>>>>                     that out!
>>>>>>
>>>>>>                 for sake of public knowledge, just to add on time
>>>>>>                 specific features: there is also $timef(format)
>>>>>>                 which returns current time attributes based on
>>>>>>                 strftime specifiers -- its documentation was
>>>>>>                 missing, I just added it. Also, there is a
>>>>>>                 transformation {s.ftime,format) which can take
>>>>>>                 any integer variable holding timestamp and return
>>>>>>                 value based on strftime format.
>>>>>>
>>>>>>
>>>>>>                 Cheers,
>>>>>>                 Daniel
>>>>>>
>>>>>>                 -- 
>>>>>>                 Daniel-Constantin Mierla -- http://www.asipto.com
>>>>>>                 http://linkedin.com/in/miconda --
>>>>>>                 http://twitter.com/miconda
>>>>>>
>>>>>>
>>>>>>                 _______________________________________________
>>>>>>                 SIP Express Router (SER) and Kamailio (OpenSER) -
>>>>>>                 sr-users mailing list
>>>>>>                 sr-users at lists.sip-router.org
>>>>>>                 <mailto:sr-users at lists.sip-router.org>
>>>>>>                 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>>>
>>>>>
>>>>>             -- 
>>>>>             Daniel-Constantin Mierla -http://www.asipto.com
>>>>>             http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>>>             Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>>>             Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>>
>>>             -- 
>>>             Daniel-Constantin Mierla -http://www.asipto.com
>>>             http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>             Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>             Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>>
>>>
>>
>>         -- 
>>         Daniel-Constantin Mierla -http://www.asipto.com
>>         http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>         Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>         Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>     Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From brandon at cryy.com  Tue Aug 14 22:23:35 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 13:23:35 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502AB36D.7080006@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
	<CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
	<502AB0E5.10303@gmail.com>
	<CACr1DbDt2N5SAYLZF40GF7FAHDAzc_yLVbCP+f=SvEdWm1955w@mail.gmail.com>
	<502AB36D.7080006@gmail.com>
Message-ID: <CACr1DbB16kz2_Ruw2M0suECO3ng+0dBkp0169-ay-j0vVADjxw@mail.gmail.com>

WORKS!

Thank you.

On Tue, Aug 14, 2012 at 1:22 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> the $timef(...) is evaluated, but you don't have time specifiers there,
> only static letters. You have to use % in front of the letters, as I guess
> from the example:
>
> http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables
>
> Try acc_$timef(%Y%m%d).
>
> Cheers,
> Daniel
>
>
>
> On 8/14/12 10:17 PM, Brandon Armstead wrote:
>
> Daniel,
>
>      My apologies - forgot to make clean.  However still no cigar.
>
>  Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
> [km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' doesn't
> exist
> Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc [acc.c:405]:
> failed to insert into database
>
>  Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> are you sure you re-installed and the right acc.so is used?
>>
>> The support is for generic PV, if it works with one it should work with
>> any. Which one does work for you?
>>
>> Can you give exact table name as printed in the SQL query? Does it have
>> the parenthesis and the format string?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/14/12 9:55 PM, Brandon Armstead wrote:
>>
>> Daniel,
>>
>>     Patch applies fine - it still does not seem to take when using
>> $timef, i..e
>>
>>  modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>>
>>  It looks as if the $timef is not being interpreted and is simply
>> writing to acc'ing i.e. INSERT INTO acc_$timef
>>
>>  Thanks!
>>
>>  Sincerely,
>> Brandon Armstead
>> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> I reapplied the patch (cherry-picked from the initial one) to the master
>>> branch.
>>>
>>> Can you test that and see if it works fine?
>>>
>>> You can cherry-picked to your branch, try:
>>>
>>> git pull origin
>>> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>>
>>> If it applies fine, compiles and the test go ok, then I will backport as
>>> soon as Juha confirms that was no solid reason in discarding this feature
>>> by his commit.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>>
>>> Daniel,
>>>
>>>     In my research I saw that commit as well but figured it was some
>>> kind of weird merging error.  Thanks for your time!  Look forward to
>>> hearing back from you guys.
>>>
>>>  Sincerely,
>>> Brandon Armstead
>>>
>>> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
>>> miconda at gmail.com> wrote:
>>>
>>>>  Hello,
>>>>
>>>> checked the sources and it seems that Juha reverted this feature with
>>>> the commit:
>>>>
>>>>
>>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>>
>>>> I guess it was accidentally, I will ask on devels list.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>>
>>>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>>
>>>> Let me also add that I am using 3.2 but I see the commit was quite some
>>>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <
>>>> miconda at gmail.com> wrote:
>>>>
>>>>   Hello,
>>>>
>>>> this functionality should be already there. Doesn't work for you?
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>
>>>> Sorry to wake up an old thread.
>>>>
>>>>  However - I am looking to export this $timef function to the param
>>>> initialization for accounting, i.e. db_table_acc
>>>>
>>>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>
>>>>  I've looked into completing this myself however I simply am not
>>>> familiar enough at this point between the three different modules that it
>>>> would take to implement this (acc, dbsr1, pv).
>>>>
>>>>  Look forward to any help / insight you may be able to provide.
>>>>
>>>>  Thanks as always!
>>>>
>>>>  Sincerely,
>>>> Brandon Armstead
>>>>
>>>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
>>>> miconda at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>>
>>>>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>>
>>>>>> Hi Daniel,
>>>>>>
>>>>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>>>
>>>>>>> I don't know what are all the functions you think of, but for the
>>>>>>> example provided above, config file does it easy right now. There is
>>>>>>> a
>>>>>>> pseudo-variable that gives broken-time attribute that can be used
>>>>>>> with
>>>>>>> avp_check(), iirc, should be:
>>>>>>>
>>>>>>> avp_db_load(...);
>>>>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>>>
>>>>>>> Of course there is the option of doing while loop, but maybe gets to
>>>>>>> large for desired config file.
>>>>>>>
>>>>>> This is pretty much what I had in mind with my new functions/module,
>>>>>> but
>>>>>> I've completely overlooked that PV when searching the docs for this
>>>>>> feature. Thank you very much for pointing that out!
>>>>>>
>>>>>  for sake of public knowledge, just to add on time specific features:
>>>>> there is also $timef(format) which returns current time attributes based on
>>>>> strftime specifiers -- its documentation was missing, I just added it.
>>>>> Also, there is a transformation {s.ftime,format) which can take any integer
>>>>> variable holding timestamp and return value based on strftime format.
>>>>>
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> --
>>>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>>>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>>
>>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From brandon at cryy.com  Tue Aug 14 23:14:56 2012
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 14 Aug 2012 14:14:56 -0700
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <502AB36D.7080006@gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
	<CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
	<502AB0E5.10303@gmail.com>
	<CACr1DbDt2N5SAYLZF40GF7FAHDAzc_yLVbCP+f=SvEdWm1955w@mail.gmail.com>
	<502AB36D.7080006@gmail.com>
Message-ID: <CACr1DbDYVaVkUys41aDo+1mK+vDxP9TQJ6LRH1eVzfQaGAy6Fg@mail.gmail.com>

Daniel,

   One question I do have - is the $timef(%Y%m%d)  evaluated only once at
run time (when kamailio is started) - or is it evaluated at the time of the
ACC'ing action.

Sincerely,
Brandon Armstead

On Tue, Aug 14, 2012 at 1:22 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> the $timef(...) is evaluated, but you don't have time specifiers there,
> only static letters. You have to use % in front of the letters, as I guess
> from the example:
>
> http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables
>
> Try acc_$timef(%Y%m%d).
>
> Cheers,
> Daniel
>
>
>
> On 8/14/12 10:17 PM, Brandon Armstead wrote:
>
> Daniel,
>
>      My apologies - forgot to make clean.  However still no cigar.
>
>  Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
> [km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' doesn't
> exist
> Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc [acc.c:405]:
> failed to insert into database
>
>  Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> are you sure you re-installed and the right acc.so is used?
>>
>> The support is for generic PV, if it works with one it should work with
>> any. Which one does work for you?
>>
>> Can you give exact table name as printed in the SQL query? Does it have
>> the parenthesis and the format string?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/14/12 9:55 PM, Brandon Armstead wrote:
>>
>> Daniel,
>>
>>     Patch applies fine - it still does not seem to take when using
>> $timef, i..e
>>
>>  modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>>
>>  It looks as if the $timef is not being interpreted and is simply
>> writing to acc'ing i.e. INSERT INTO acc_$timef
>>
>>  Thanks!
>>
>>  Sincerely,
>> Brandon Armstead
>> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> I reapplied the patch (cherry-picked from the initial one) to the master
>>> branch.
>>>
>>> Can you test that and see if it works fine?
>>>
>>> You can cherry-picked to your branch, try:
>>>
>>> git pull origin
>>> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>>
>>> If it applies fine, compiles and the test go ok, then I will backport as
>>> soon as Juha confirms that was no solid reason in discarding this feature
>>> by his commit.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>>
>>> Daniel,
>>>
>>>     In my research I saw that commit as well but figured it was some
>>> kind of weird merging error.  Thanks for your time!  Look forward to
>>> hearing back from you guys.
>>>
>>>  Sincerely,
>>> Brandon Armstead
>>>
>>> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
>>> miconda at gmail.com> wrote:
>>>
>>>>  Hello,
>>>>
>>>> checked the sources and it seems that Juha reverted this feature with
>>>> the commit:
>>>>
>>>>
>>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>>
>>>> I guess it was accidentally, I will ask on devels list.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>>
>>>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>>
>>>> Let me also add that I am using 3.2 but I see the commit was quite some
>>>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <
>>>> miconda at gmail.com> wrote:
>>>>
>>>>   Hello,
>>>>
>>>> this functionality should be already there. Doesn't work for you?
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>
>>>> Sorry to wake up an old thread.
>>>>
>>>>  However - I am looking to export this $timef function to the param
>>>> initialization for accounting, i.e. db_table_acc
>>>>
>>>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>
>>>>  I've looked into completing this myself however I simply am not
>>>> familiar enough at this point between the three different modules that it
>>>> would take to implement this (acc, dbsr1, pv).
>>>>
>>>>  Look forward to any help / insight you may be able to provide.
>>>>
>>>>  Thanks as always!
>>>>
>>>>  Sincerely,
>>>> Brandon Armstead
>>>>
>>>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
>>>> miconda at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>>
>>>>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>>
>>>>>> Hi Daniel,
>>>>>>
>>>>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>>>
>>>>>>> I don't know what are all the functions you think of, but for the
>>>>>>> example provided above, config file does it easy right now. There is
>>>>>>> a
>>>>>>> pseudo-variable that gives broken-time attribute that can be used
>>>>>>> with
>>>>>>> avp_check(), iirc, should be:
>>>>>>>
>>>>>>> avp_db_load(...);
>>>>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>>>
>>>>>>> Of course there is the option of doing while loop, but maybe gets to
>>>>>>> large for desired config file.
>>>>>>>
>>>>>> This is pretty much what I had in mind with my new functions/module,
>>>>>> but
>>>>>> I've completely overlooked that PV when searching the docs for this
>>>>>> feature. Thank you very much for pointing that out!
>>>>>>
>>>>>  for sake of public knowledge, just to add on time specific features:
>>>>> there is also $timef(format) which returns current time attributes based on
>>>>> strftime specifiers -- its documentation was missing, I just added it.
>>>>> Also, there is a transformation {s.ftime,format) which can take any integer
>>>>> variable holding timestamp and return value based on strftime format.
>>>>>
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> --
>>>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>>>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>>
>>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From andre at rodnoe.tv  Tue Aug 14 11:31:34 2012
From: andre at rodnoe.tv (andre)
Date: Tue, 14 Aug 2012 12:31:34 +0300
Subject: [SR-Users] centos (fedora) service startup file
In-Reply-To: <502A18B4.8040207@gmail.com>
References: <501AA9D6.4040500@rodnoe.tv> <502A18B4.8040207@gmail.com>
Message-ID: <502A1AF6.2000408@rodnoe.tv>

14.08.2012 12:21, Daniel-Constantin Mierla ?????:
> Hello,
>
> thanks, it was committed in git master branch, backports to follow 
> before minor releases.
>
> Do you build your own rpms or you install from the opensuse build system?
 From the opensuse build system.
>
> Cheers,
> Daniel
>
> On 8/2/12 6:24 PM, andre wrote:
>> Somwhere losted since openser 1.0 :)
>>
>>
>> Please fix in:
>>
>> start() {
>> echo -n $"Starting $prog: "
>> .....
>> [ $RETVAL = 0 ] && touch something
>> to
>> [ $RETVAL = 0 ] && touch someting && success
>>
>> I mean call function
>> success
>> in the end of the expression.
>>
>> Thanks in advance
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From bligthart at btlnet.co.uk  Wed Aug 15 09:31:27 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Wed, 15 Aug 2012 07:31:27 +0000
Subject: [SR-Users] db_cassandra
In-Reply-To: <502A5829.3090201@1and1.ro>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A46FE.7080403@1and1.ro>
	<D8261FE064C78B44B77EF7C4569D6CBF0D11AF60@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A5829.3090201@1and1.ro>
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11B39B@DB3PRD0510MB380.eurprd05.prod.outlook.com>

Thanks again, Anca

Bou
From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Anca Vamanu
Sent: 14 August 2012 14:53
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] db_cassandra

Hi,


If you are using cassandra-cli you can fill in the tables using something like this:

version table:

set version['location']['table_version']='1004';
set version['subscriber']['table_version']='6';

and the subscriber table:

set subscriber['userX']['domain']='domain.com';
set subscriber['userX']['password']='pass123';


Regards,
Anca


On 08/14/2012 04:26 PM, Boudewyn Ligthart wrote:
HI Anca,

Thanks for pointing me to the latest GIT version, this compiles as per normal.

Another question,

Have you got pointers on how to install the tables / setup Cassandra.
I have created the column families for location, subscriber, version but not yet filled these with data.

Regards

Bou

From: sr-users-bounces at lists.sip-router.org<mailto:sr-users-bounces at lists.sip-router.org> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Anca Vamanu
Sent: 14 August 2012 13:39
To: sr-users at lists.sip-router.org<mailto:sr-users at lists.sip-router.org>
Subject: Re: [SR-Users] db_cassandra

Hi,


I tried compiling the module with the latest 3.1 from git and indeed it did not compile initially.
This very simple patch that is attached was needed. There were some problems with implicit casts.

I see that you are talking about 3.1.1 version, I have not tried with this exact version. You should maybe consider using the latest git for 3.1, it should also be more stable.


Regards,
Anca





On 08/14/2012 01:05 PM, Boudewyn Ligthart wrote:
Hi,

I am trying to get the db_cassandra working in kamailio 3.1.1 with thrift 0.6.1 API and however I get the following
compile error.

ol.h:33,
                 from dbcassa_base.h:38,
                 from dbcassa_base.cpp:47:
../../lib/srdb1/db.h:388:20: error: expected unqualified-id before ?delete?
dbcassa_base.cpp: In function ?ColumnVecPtr cassa_translate_query(const db1_con_t*, str* const*, const db_val_t*, str* const*, int, int, int*)?:
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 7 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 5 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 6 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 4 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
dbcassa_base.cpp:538:8: warning: format ?%d? expects argument of type ?int?, but argument 4 has type ?std::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}? [-Wformat]
../../lib/srdb1/db_res.h: At global scope:
../../lib/srdb1/db_res.h:105:19: warning: inline function ?db1_res_t* db_new_result()? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:121:12: warning: inline function ?int db_allocate_columns(db1_res_t*, unsigned int)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:98:12: warning: inline function ?int db_free_columns(db1_res_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:129:12: warning: inline function ?int db_allocate_rows(db1_res_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_row.h:74:12: warning: inline function ?int db_allocate_row(const db1_res_t*, db_row_t*)? used but never defined [enabled by default]
../../lib/srdb1/db_res.h:111:12: warning: inline function ?int db_free_result(db1_res_t*)? used but never defined [enabled by default]
make[1]: *** [dbcassa_base.o] Error 1
make: *** [install-modules] Error 1


Anybody has got any ideas?

Thanks in advance,

Bou



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From danb.lists at gmail.com  Wed Aug 15 11:30:52 2012
From: danb.lists at gmail.com (DanB)
Date: Wed, 15 Aug 2012 11:30:52 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
	grp_avp
In-Reply-To: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
Message-ID: <502B6C4C.2020609@gmail.com>

Hey Guys,

Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.

Having the following code:
"""
if ds_is_from_list() {
   xlog("Request coming from SCloud");
   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
}
"""

will produce the following output to syslog (dsgrp null instead of group 
id):
"""
Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: 
Request coming from SCloud
Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: 
Checking grp_internal, having: <null>
"""
I have in dispatcher set:
modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
and I am loading the rules from file.

I should mention that this block was working fine before upgrade.

Am I doing something wrong?

Thanks in advance!
DanB



From miconda at gmail.com  Wed Aug 15 11:41:56 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Aug 2012 11:41:56 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B6C4C.2020609@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com>
Message-ID: <502B6EE4.8090508@gmail.com>

Hello,

there was no change in dispatcher module related to this function since 
3.3.0. The value of the group id is set to the variable specified by the 
parameter setid_pvar:

http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800

Very likely you are using the wrong variable which you set as grp_avp 
parameter, based your config snippets.

Cheers,
Daniel

On 8/15/12 11:30 AM, DanB wrote:
> Hey Guys,
>
> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>
> Having the following code:
> """
> if ds_is_from_list() {
>   xlog("Request coming from SCloud");
>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
> }
> """
>
> will produce the following output to syslog (dsgrp null instead of 
> group id):
> """
> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: 
> Request coming from SCloud
> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: 
> Checking grp_internal, having: <null>
> """
> I have in dispatcher set:
> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
> and I am loading the rules from file.
>
> I should mention that this block was working fine before upgrade.
>
> Am I doing something wrong?
>
> Thanks in advance!
> DanB
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From danb.lists at gmail.com  Wed Aug 15 12:37:14 2012
From: danb.lists at gmail.com (DanB)
Date: Wed, 15 Aug 2012 12:37:14 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B6EE4.8090508@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
Message-ID: <502B7BDA.4090802@gmail.com>

Hey Daniel,

Thanks for so fast feedback. I have doublecked again the config:

If it helps, here are my complete settings:
"""
# ----- dispatcher params -----
modparam("dispatcher", "list_file", "/etc/sbalancer/dispatcher.list")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(dsdst)")
modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
modparam("dispatcher", "cnt_avp", "$avp(dscnt)")
modparam("dispatcher", "attrs_avp", "$avp(dsattrs)")
modparam("dispatcher", "hash_pvar", "$avp(dshash)")
modparam("dispatcher", "ds_ping_from", "sip:SBalancer at mydomain.com")
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dispatcher", "ds_probing_threshhold", 3)
modparam("dispatcher", "ds_ping_reply_codes", 
"class=2;code=403;code=488;class=3")
modparam("dispatcher", "ds_probing_mode", 1)
"""

Calling the logging block happens right at beginning, so not possible to 
overwrite the avp.
"""
request_route {
	#check antiflood before anything
	route(ANTIFLOOD);

	# Detect if requests are from internal network
	if ds_is_from_list() {
		xlog("Request coming from SCloud");
		$avp(s:authType) = "INTERNAL";
		$avp(s:authUser) = $hdr(X-AuthUser);
		xlog("Checking grp_internal, having: $avp(dsgrp)");
	}
"""

Result the same ( I am sure ds_is_from_list matches since I get the 
first xlog correctly):
"""
Aug 15 12:36:07 PrxyDev /usr/sbin/kamailio[2973]: ERROR: <script>: 
Request coming from SCloud
Aug 15 12:36:07 PrxyDev /usr/sbin/kamailio[2973]: ERROR: <script>: 
Checking grp_internal, having: <null>
"""

Thanks again,
DanB




On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> there was no change in dispatcher module related to this function since
> 3.3.0. The value of the group id is set to the variable specified by the
> parameter setid_pvar:
>
> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800
>
> Very likely you are using the wrong variable which you set as grp_avp
> parameter, based your config snippets.
>
> Cheers,
> Daniel
>
> On 8/15/12 11:30 AM, DanB wrote:
>> Hey Guys,
>>
>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>
>> Having the following code:
>> """
>> if ds_is_from_list() {
>>   xlog("Request coming from SCloud");
>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>> }
>> """
>>
>> will produce the following output to syslog (dsgrp null instead of
>> group id):
>> """
>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>> Request coming from SCloud
>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>> Checking grp_internal, having: <null>
>> """
>> I have in dispatcher set:
>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>> and I am loading the rules from file.
>>
>> I should mention that this block was working fine before upgrade.
>>
>> Am I doing something wrong?
>>
>> Thanks in advance!
>> DanB
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From danb.lists at gmail.com  Wed Aug 15 12:40:33 2012
From: danb.lists at gmail.com (DanB)
Date: Wed, 15 Aug 2012 12:40:33 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B6EE4.8090508@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
Message-ID: <502B7CA1.6080907@gmail.com>

Daniel,

My bad, I was reading through the lines :(.

I now understand that setid_pvar is different than group pvar. I got the 
false understanding based on documentation saying:
"""
Parameter groupid is optional, when it is missing, then the matching 
will be done against all addresses in all groups. Upon a match, the 
'grp_avp' will be set to groupid of matching address.
"""

All clear now, thanks!!!

DanB

On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> there was no change in dispatcher module related to this function since
> 3.3.0. The value of the group id is set to the variable specified by the
> parameter setid_pvar:
>
> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800
>
> Very likely you are using the wrong variable which you set as grp_avp
> parameter, based your config snippets.
>
> Cheers,
> Daniel
>
> On 8/15/12 11:30 AM, DanB wrote:
>> Hey Guys,
>>
>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>
>> Having the following code:
>> """
>> if ds_is_from_list() {
>>   xlog("Request coming from SCloud");
>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>> }
>> """
>>
>> will produce the following output to syslog (dsgrp null instead of
>> group id):
>> """
>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>> Request coming from SCloud
>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>> Checking grp_internal, having: <null>
>> """
>> I have in dispatcher set:
>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>> and I am loading the rules from file.
>>
>> I should mention that this block was working fine before upgrade.
>>
>> Am I doing something wrong?
>>
>> Thanks in advance!
>> DanB
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From miconda at gmail.com  Wed Aug 15 12:52:04 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Aug 2012 12:52:04 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B7CA1.6080907@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
	<502B7CA1.6080907@gmail.com>
Message-ID: <502B7F54.4010808@gmail.com>

Hello,

right, the readme was mistaken. Probably grp_avp was used initially but 
changed later because it messes up when ds_is_from_list() and 
ds_select_*() are used for same requests.

I committed an update to the readme.

Cheers,
Daniel

On 8/15/12 12:40 PM, DanB wrote:
> Daniel,
>
> My bad, I was reading through the lines :(.
>
> I now understand that setid_pvar is different than group pvar. I got 
> the false understanding based on documentation saying:
> """
> Parameter groupid is optional, when it is missing, then the matching 
> will be done against all addresses in all groups. Upon a match, the 
> 'grp_avp' will be set to groupid of matching address.
> """
>
> All clear now, thanks!!!
>
> DanB
>
> On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> there was no change in dispatcher module related to this function since
>> 3.3.0. The value of the group id is set to the variable specified by the
>> parameter setid_pvar:
>>
>> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800 
>>
>>
>> Very likely you are using the wrong variable which you set as grp_avp
>> parameter, based your config snippets.
>>
>> Cheers,
>> Daniel
>>
>> On 8/15/12 11:30 AM, DanB wrote:
>>> Hey Guys,
>>>
>>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>>
>>> Having the following code:
>>> """
>>> if ds_is_from_list() {
>>>   xlog("Request coming from SCloud");
>>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>>> }
>>> """
>>>
>>> will produce the following output to syslog (dsgrp null instead of
>>> group id):
>>> """
>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>> Request coming from SCloud
>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>> Checking grp_internal, having: <null>
>>> """
>>> I have in dispatcher set:
>>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>>> and I am loading the rules from file.
>>>
>>> I should mention that this block was working fine before upgrade.
>>>
>>> Am I doing something wrong?
>>>
>>> Thanks in advance!
>>> DanB
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From miconda at gmail.com  Wed Aug 15 12:53:24 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Aug 2012 12:53:24 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B7BDA.4090802@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
	<502B7BDA.4090802@gmail.com>
Message-ID: <502B7FA4.1010809@gmail.com>

Hello,

grp_avp is set only when calling the destination/domain select functions.

You have to set the parameter setid_pvar:

http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800

And use the variable you set for this parameter after ds_is_from_list().

So, there are two parameters, grp_avp and setid_pvar.

Cheers,
Daniel

On 8/15/12 12:37 PM, DanB wrote:
> Hey Daniel,
>
> Thanks for so fast feedback. I have doublecked again the config:
>
> If it helps, here are my complete settings:
> """
> # ----- dispatcher params -----
> modparam("dispatcher", "list_file", "/etc/sbalancer/dispatcher.list")
> modparam("dispatcher", "flags", 2)
> modparam("dispatcher", "dst_avp", "$avp(dsdst)")
> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
> modparam("dispatcher", "cnt_avp", "$avp(dscnt)")
> modparam("dispatcher", "attrs_avp", "$avp(dsattrs)")
> modparam("dispatcher", "hash_pvar", "$avp(dshash)")
> modparam("dispatcher", "ds_ping_from", "sip:SBalancer at mydomain.com")
> modparam("dispatcher", "ds_ping_interval", 30)
> modparam("dispatcher", "ds_probing_threshhold", 3)
> modparam("dispatcher", "ds_ping_reply_codes", 
> "class=2;code=403;code=488;class=3")
> modparam("dispatcher", "ds_probing_mode", 1)
> """
>
> Calling the logging block happens right at beginning, so not possible 
> to overwrite the avp.
> """
> request_route {
>     #check antiflood before anything
>     route(ANTIFLOOD);
>
>     # Detect if requests are from internal network
>     if ds_is_from_list() {
>         xlog("Request coming from SCloud");
>         $avp(s:authType) = "INTERNAL";
>         $avp(s:authUser) = $hdr(X-AuthUser);
>         xlog("Checking grp_internal, having: $avp(dsgrp)");
>     }
> """
>
> Result the same ( I am sure ds_is_from_list matches since I get the 
> first xlog correctly):
> """
> Aug 15 12:36:07 PrxyDev /usr/sbin/kamailio[2973]: ERROR: <script>: 
> Request coming from SCloud
> Aug 15 12:36:07 PrxyDev /usr/sbin/kamailio[2973]: ERROR: <script>: 
> Checking grp_internal, having: <null>
> """
>
> Thanks again,
> DanB
>
>
>
>
> On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> there was no change in dispatcher module related to this function since
>> 3.3.0. The value of the group id is set to the variable specified by the
>> parameter setid_pvar:
>>
>> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800 
>>
>>
>> Very likely you are using the wrong variable which you set as grp_avp
>> parameter, based your config snippets.
>>
>> Cheers,
>> Daniel
>>
>> On 8/15/12 11:30 AM, DanB wrote:
>>> Hey Guys,
>>>
>>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>>
>>> Having the following code:
>>> """
>>> if ds_is_from_list() {
>>>   xlog("Request coming from SCloud");
>>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>>> }
>>> """
>>>
>>> will produce the following output to syslog (dsgrp null instead of
>>> group id):
>>> """
>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>> Request coming from SCloud
>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>> Checking grp_internal, having: <null>
>>> """
>>> I have in dispatcher set:
>>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>>> and I am loading the rules from file.
>>>
>>> I should mention that this block was working fine before upgrade.
>>>
>>> Am I doing something wrong?
>>>
>>> Thanks in advance!
>>> DanB
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From danb.lists at gmail.com  Wed Aug 15 12:56:33 2012
From: danb.lists at gmail.com (DanB)
Date: Wed, 15 Aug 2012 12:56:33 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B7F54.4010808@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
	<502B7CA1.6080907@gmail.com> <502B7F54.4010808@gmail.com>
Message-ID: <502B8061.1050308@gmail.com>

Daniel,

Since we are still in "connected" state, one more (hopefully last) 
question on this subject:

If I set the "setid_pvar" the configuration does not longer pass 
checking (just upgraded to latest nightly 3.3.1):
"""
root at PrxyDev:/etc/sbalancer# kamailio -c -f kamailio.cfg
loading modules under 
/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/local/lib/kamailio/modules/
  0(3190) ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter 
<setid_pvar> not found in module <dispatcher>
  0(3190) : <core> [cfg.y:3594]: parse error in config file 
/usr/local/src/sipean2/trunk/SDev/SBalancer/Kamailio/etc/kamailio/kamailio.cfg, 
line 73, column 51: Can't set module parameter
ERROR: bad config file (1 errors)
root at PrxyDev:/etc/sbalancer# kamailio -V
version: kamailio 3.3.1 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, 
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 
F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, 
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 03:07:36 Aug 15 2012 with gcc 4.4.5
"""

I define the parameter in the following way:
"""
modparam("dispatcher", "setid_pvar", "$var(setid)")
"""

Am I missing something again?


Thanks,
DanB

On 08/15/2012 12:52 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> right, the readme was mistaken. Probably grp_avp was used initially but
> changed later because it messes up when ds_is_from_list() and
> ds_select_*() are used for same requests.
>
> I committed an update to the readme.
>
> Cheers,
> Daniel
>
> On 8/15/12 12:40 PM, DanB wrote:
>> Daniel,
>>
>> My bad, I was reading through the lines :(.
>>
>> I now understand that setid_pvar is different than group pvar. I got
>> the false understanding based on documentation saying:
>> """
>> Parameter groupid is optional, when it is missing, then the matching
>> will be done against all addresses in all groups. Upon a match, the
>> 'grp_avp' will be set to groupid of matching address.
>> """
>>
>> All clear now, thanks!!!
>>
>> DanB
>>
>> On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> there was no change in dispatcher module related to this function since
>>> 3.3.0. The value of the group id is set to the variable specified by the
>>> parameter setid_pvar:
>>>
>>> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800
>>>
>>>
>>> Very likely you are using the wrong variable which you set as grp_avp
>>> parameter, based your config snippets.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 8/15/12 11:30 AM, DanB wrote:
>>>> Hey Guys,
>>>>
>>>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>>>
>>>> Having the following code:
>>>> """
>>>> if ds_is_from_list() {
>>>>   xlog("Request coming from SCloud");
>>>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>>>> }
>>>> """
>>>>
>>>> will produce the following output to syslog (dsgrp null instead of
>>>> group id):
>>>> """
>>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>>> Request coming from SCloud
>>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>>> Checking grp_internal, having: <null>
>>>> """
>>>> I have in dispatcher set:
>>>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>>>> and I am loading the rules from file.
>>>>
>>>> I should mention that this block was working fine before upgrade.
>>>>
>>>> Am I doing something wrong?
>>>>
>>>> Thanks in advance!
>>>> DanB
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>



From miconda at gmail.com  Wed Aug 15 13:00:12 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Aug 2012 13:00:12 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B8061.1050308@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
	<502B7CA1.6080907@gmail.com> <502B7F54.4010808@gmail.com>
	<502B8061.1050308@gmail.com>
Message-ID: <502B813C.4030507@gmail.com>

Another mismatch of sources and docs, the parameter is expected to be 
'setid_pvname' instead of 'setid_pvar'. I'll update docs.

Cheers,
Daniel

On 8/15/12 12:56 PM, DanB wrote:
> Daniel,
>
> Since we are still in "connected" state, one more (hopefully last) 
> question on this subject:
>
> If I set the "setid_pvar" the configuration does not longer pass 
> checking (just upgraded to latest nightly 3.3.1):
> """
> root at PrxyDev:/etc/sbalancer# kamailio -c -f kamailio.cfg
> loading modules under 
> /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/local/lib/kamailio/modules/
>  0(3190) ERROR: <core> [modparam.c:151]: set_mod_param_regex: 
> parameter <setid_pvar> not found in module <dispatcher>
>  0(3190) : <core> [cfg.y:3594]: parse error in config file 
> /usr/local/src/sipean2/trunk/SDev/SBalancer/Kamailio/etc/kamailio/kamailio.cfg, 
> line 73, column 51: Can't set module parameter
> ERROR: bad config file (1 errors)
> root at PrxyDev:/etc/sbalancer# kamailio -V
> version: kamailio 3.3.1 (i386/linux)
> flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, 
> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 
> SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, 
> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, 
> HAVE_RESOLV_RES
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> id: unknown
> compiled on 03:07:36 Aug 15 2012 with gcc 4.4.5
> """
>
> I define the parameter in the following way:
> """
> modparam("dispatcher", "setid_pvar", "$var(setid)")
> """
>
> Am I missing something again?
>
>
> Thanks,
> DanB
>
> On 08/15/2012 12:52 PM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> right, the readme was mistaken. Probably grp_avp was used initially but
>> changed later because it messes up when ds_is_from_list() and
>> ds_select_*() are used for same requests.
>>
>> I committed an update to the readme.
>>
>> Cheers,
>> Daniel
>>
>> On 8/15/12 12:40 PM, DanB wrote:
>>> Daniel,
>>>
>>> My bad, I was reading through the lines :(.
>>>
>>> I now understand that setid_pvar is different than group pvar. I got
>>> the false understanding based on documentation saying:
>>> """
>>> Parameter groupid is optional, when it is missing, then the matching
>>> will be done against all addresses in all groups. Upon a match, the
>>> 'grp_avp' will be set to groupid of matching address.
>>> """
>>>
>>> All clear now, thanks!!!
>>>
>>> DanB
>>>
>>> On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
>>>> Hello,
>>>>
>>>> there was no change in dispatcher module related to this function 
>>>> since
>>>> 3.3.0. The value of the group id is set to the variable specified 
>>>> by the
>>>> parameter setid_pvar:
>>>>
>>>> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800 
>>>>
>>>>
>>>>
>>>> Very likely you are using the wrong variable which you set as grp_avp
>>>> parameter, based your config snippets.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/15/12 11:30 AM, DanB wrote:
>>>>> Hey Guys,
>>>>>
>>>>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>>>>
>>>>> Having the following code:
>>>>> """
>>>>> if ds_is_from_list() {
>>>>>   xlog("Request coming from SCloud");
>>>>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>>>>> }
>>>>> """
>>>>>
>>>>> will produce the following output to syslog (dsgrp null instead of
>>>>> group id):
>>>>> """
>>>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>>>> Request coming from SCloud
>>>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>>>> Checking grp_internal, having: <null>
>>>>> """
>>>>> I have in dispatcher set:
>>>>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>>>>> and I am loading the rules from file.
>>>>>
>>>>> I should mention that this block was working fine before upgrade.
>>>>>
>>>>> Am I doing something wrong?
>>>>>
>>>>> Thanks in advance!
>>>>> DanB
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>>>> list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From danb.lists at gmail.com  Wed Aug 15 13:02:52 2012
From: danb.lists at gmail.com (DanB)
Date: Wed, 15 Aug 2012 13:02:52 +0200
Subject: [SR-Users] Dispatcher: ds_is_from_list does not longer populate
 grp_avp
In-Reply-To: <502B813C.4030507@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502B6C4C.2020609@gmail.com> <502B6EE4.8090508@gmail.com>
	<502B7CA1.6080907@gmail.com> <502B7F54.4010808@gmail.com>
	<502B8061.1050308@gmail.com> <502B813C.4030507@gmail.com>
Message-ID: <502B81DC.8030206@gmail.com>

Daniel,

Nice catch! All smooth now.

Thanks and enjoy your free time ;).

DanB

On 08/15/2012 01:00 PM, Daniel-Constantin Mierla wrote:
> Another mismatch of sources and docs, the parameter is expected to be
> 'setid_pvname' instead of 'setid_pvar'. I'll update docs.
>
> Cheers,
> Daniel
>
> On 8/15/12 12:56 PM, DanB wrote:
>> Daniel,
>>
>> Since we are still in "connected" state, one more (hopefully last)
>> question on this subject:
>>
>> If I set the "setid_pvar" the configuration does not longer pass
>> checking (just upgraded to latest nightly 3.3.1):
>> """
>> root at PrxyDev:/etc/sbalancer# kamailio -c -f kamailio.cfg
>> loading modules under
>> /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/local/lib/kamailio/modules/
>>
>>  0(3190) ERROR: <core> [modparam.c:151]: set_mod_param_regex:
>> parameter <setid_pvar> not found in module <dispatcher>
>>  0(3190) : <core> [cfg.y:3594]: parse error in config file
>> /usr/local/src/sipean2/trunk/SDev/SBalancer/Kamailio/etc/kamailio/kamailio.cfg,
>> line 73, column 51: Can't set module parameter
>> ERROR: bad config file (1 errors)
>> root at PrxyDev:/etc/sbalancer# kamailio -V
>> version: kamailio 3.3.1 (i386/linux)
>> flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS,
>> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM,
>> SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
>> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
>> HAVE_RESOLV_RES
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>> id: unknown
>> compiled on 03:07:36 Aug 15 2012 with gcc 4.4.5
>> """
>>
>> I define the parameter in the following way:
>> """
>> modparam("dispatcher", "setid_pvar", "$var(setid)")
>> """
>>
>> Am I missing something again?
>>
>>
>> Thanks,
>> DanB
>>
>> On 08/15/2012 12:52 PM, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> right, the readme was mistaken. Probably grp_avp was used initially but
>>> changed later because it messes up when ds_is_from_list() and
>>> ds_select_*() are used for same requests.
>>>
>>> I committed an update to the readme.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 8/15/12 12:40 PM, DanB wrote:
>>>> Daniel,
>>>>
>>>> My bad, I was reading through the lines :(.
>>>>
>>>> I now understand that setid_pvar is different than group pvar. I got
>>>> the false understanding based on documentation saying:
>>>> """
>>>> Parameter groupid is optional, when it is missing, then the matching
>>>> will be done against all addresses in all groups. Upon a match, the
>>>> 'grp_avp' will be set to groupid of matching address.
>>>> """
>>>>
>>>> All clear now, thanks!!!
>>>>
>>>> DanB
>>>>
>>>> On 08/15/2012 11:41 AM, Daniel-Constantin Mierla wrote:
>>>>> Hello,
>>>>>
>>>>> there was no change in dispatcher module related to this function
>>>>> since
>>>>> 3.3.0. The value of the group id is set to the variable specified
>>>>> by the
>>>>> parameter setid_pvar:
>>>>>
>>>>> http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2495800
>>>>>
>>>>>
>>>>>
>>>>> Very likely you are using the wrong variable which you set as grp_avp
>>>>> parameter, based your config snippets.
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> On 8/15/12 11:30 AM, DanB wrote:
>>>>>> Hey Guys,
>>>>>>
>>>>>> Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1.
>>>>>>
>>>>>> Having the following code:
>>>>>> """
>>>>>> if ds_is_from_list() {
>>>>>>   xlog("Request coming from SCloud");
>>>>>>   xlog("Checking grp_internal, having: $avp(s:dsgrp)");
>>>>>> }
>>>>>> """
>>>>>>
>>>>>> will produce the following output to syslog (dsgrp null instead of
>>>>>> group id):
>>>>>> """
>>>>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>>>>> Request coming from SCloud
>>>>>> Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>:
>>>>>> Checking grp_internal, having: <null>
>>>>>> """
>>>>>> I have in dispatcher set:
>>>>>> modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
>>>>>> and I am loading the rules from file.
>>>>>>
>>>>>> I should mention that this block was working fine before upgrade.
>>>>>>
>>>>>> Am I doing something wrong?
>>>>>>
>>>>>> Thanks in advance!
>>>>>> DanB
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>>>> list
>>>>>> sr-users at lists.sip-router.org
>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>
>>
>



From Fchahrour at vanrise.com  Wed Aug 15 20:23:50 2012
From: Fchahrour at vanrise.com (Fatima Chahrour)
Date: Wed, 15 Aug 2012 21:23:50 +0300
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <06b501cd7a17$df1452e0$9d3cf8a0$@com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>
	<5029835A.4000108@evaristesys.com>
	<065801cd79ff$5b491530$11db3f90$@com>
	<502A1E9F.5070608@sipwise.com>
	<067701cd7a04$27fa37e0$77eea7a0$@com>
	<06b501cd7a17$df1452e0$9d3cf8a0$@com>
Message-ID: <A1825892-76C2-47D6-A980-A41D21D47B26@vanrise.com>

Dears,

Any help or ideas regarding the below?

Thanks,
F.Chahrour

Sent from iPhone

On Aug 14, 2012, at 3:25 PM, "Fatima Chahrour~Vanrise Support" <fchahrour at vanrise.com> wrote:

> Can I ask one more question please?!
> 
> Am planning to send the routing decision configured in kamailio to ACME SBC;
> but wanted to know if anyone has an idea about the format of this routing
> that ACME will take from Kamailio. In other words, what is the routing
> output from Kamailio and that works to be as an input for ACME? And how will
> ACME forward a call to Kamailio to get this routing?
> 
> For example let's take the following example which summarize my target:
> Call from a customer will hit the ACME SBC, and the ACME SBC which will take
> the routing choice from Kamailio. 
> 
> 
> 1- EX call to 9613555555 with Supplier A as first choice in routing and
> Supplier B as second choice with less priority (on Kamailio)
> 2- A customer on ACME calls 9613555555 where ACME will send request to
> Kamailio and get supplier A as result
> 3- ACME will try the call on supplier B (either succeeds or fails)
> 4- In case of failure, ACME should return (something!!!) to Kamailio and get
> Supplier B as second choice.
> 
> Can you explain how the above is configures between ACME (or any other SBC)
> and Kamailio?
> 
> Regards,
> F.Chahrour
> 
> -----Original Message-----
> From: sr-users-bounces at lists.sip-router.org
> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fatima
> Chahrour~Vanrise Support
> Sent: Tuesday, August 14, 2012 1:04 PM
> To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
> Mailing List'
> Subject: Re: [SR-Users] Kamailio and ACME
> 
> Thank you Andrea. 
> I just wanted to check if there is any incompatibility between both before
> surfing deeply.
> 
> -----Original Message-----
> From: sr-users-bounces at lists.sip-router.org
> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Andreas Granig
> Sent: Tuesday, August 14, 2012 12:47 PM
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] Kamailio and ACME
> 
> On 08/14/2012 11:29 AM, Fatima Chahrour~Vanrise Support wrote:
>> I wanted to ask if Kamailio is compatible with ACME Packet or 
>> experianced by anyone in the list?
>> and if there is any document that you recommend about such topic 
>> espacially in what is related to routing?
> 
> There is nothing special for interoperating Kamailio with ACME, it's treated
> like any other SIP device. As always, it depends on what you want to do.
> 
> Andreas
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


From agranig at sipwise.com  Thu Aug 16 00:43:30 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 16 Aug 2012 00:43:30 +0200
Subject: [SR-Users] Kamailio and ACME
In-Reply-To: <A1825892-76C2-47D6-A980-A41D21D47B26@vanrise.com>
References: <D2BE9A8E-2A8B-414C-9034-9BAD213863F8@vanrise.com>
	<5029835A.4000108@evaristesys.com>
	<065801cd79ff$5b491530$11db3f90$@com>
	<502A1E9F.5070608@sipwise.com>
	<067701cd7a04$27fa37e0$77eea7a0$@com>
	<06b501cd7a17$df1452e0$9d3cf8a0$@com>
	<A1825892-76C2-47D6-A980-A41D21D47B26@vanrise.com>
Message-ID: <502C2612.2080007@sipwise.com>

On 08/15/2012 08:23 PM, Fatima Chahrour wrote:
> Any help or ideas regarding the below?

Both lcr or dispatcher module can do what you want, check the docs at
http://kamailio.org/docs/modules/stable/ .

You need to get an idea how this stuff works by studying the example
config shipped with kamailio first though. Kamailio is a powerful proxy
which can do pretty much everything you want, but you need to tell it
explicitely *what* to do.

Andreas

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From phillman25 at gmail.com  Thu Aug 16 07:56:35 2012
From: phillman25 at gmail.com (phillman25)
Date: Thu, 16 Aug 2012 08:56:35 +0300
Subject: [SR-Users] rtpproxy module issue
Message-ID: <CAEprJzL2-fLfqgaLYfO5A9tBsbDPzb1WtG7-2qqHCmMUL0zzVA@mail.gmail.com>

Dear List

I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1
However, when trying to start kamailio its giving me the below error:


ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module
</usr/local/lib/kamailio/modules_k/rtpproxy.so>:
/usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc
: <core> [cfg.y:3591]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
load module


Does anyone know why rtpproxy module cannot be loaded?


This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so:



shell-init: error retrieving current directory: getcwd: cannot access
parent directories: No such file or directory
        linux-gate.so.1 =>  (0x00276000)
        libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000)
        libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000)
        libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000)
        /lib/ld-linux.so.2 (0x00114000)



Regards Phillip
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From danb.lists at gmail.com  Thu Aug 16 08:35:05 2012
From: danb.lists at gmail.com (DanB)
Date: Thu, 16 Aug 2012 08:35:05 +0200
Subject: [SR-Users] Debian packages still installing 3.3.0
In-Reply-To: <mailman.5126.1345096604.2421.sr-users@lists.sip-router.org>
References: <mailman.5126.1345096604.2421.sr-users@lists.sip-router.org>
Message-ID: <502C9499.6030801@gmail.com>

Hey Guys,

Not sure if important, but what I noticed is that the latest debian 
stable as well as latest 33 packages, they both install still kamailio 
3.3.0. In order to get 3.3.1 I needed to use nightly build packages.

Is this what is desired?

Thanks,
DanB


From manwe at aholab.ehu.es  Thu Aug 16 09:19:18 2012
From: manwe at aholab.ehu.es (Jon Bonilla (Manwe))
Date: Thu, 16 Aug 2012 09:19:18 +0200
Subject: [SR-Users] Debian packages still installing 3.3.0
In-Reply-To: <502C9499.6030801@gmail.com>
References: <mailman.5126.1345096604.2421.sr-users@lists.sip-router.org>
	<502C9499.6030801@gmail.com>
Message-ID: <20120816091918.276f94e2@multivac>

El Thu, 16 Aug 2012 08:35:05 +0200
DanB <danb.lists at gmail.com> escribi?:

> Hey Guys,
> 
> Not sure if important, but what I noticed is that the latest debian 
> stable as well as latest 33 packages, they both install still kamailio 
> 3.3.0. In order to get 3.3.1 I needed to use nightly build packages.
> 
> Is this what is desired?
> 

No. I didn't build 3.3.1 packages because I was out for vacation with no
internet access. Sorry, they will be ready this weekend. Meanwhile, as you said,
you can use a 3.3-nightly version



From khareapoorv at gmail.com  Thu Aug 16 09:40:20 2012
From: khareapoorv at gmail.com (Apoorv Khare)
Date: Thu, 16 Aug 2012 00:40:20 -0700
Subject: [SR-Users] Is Route Header in SIP Invite Mandatory?
Message-ID: <CAOfMT3O8bONUwq1HTzy92_WC89042zUmx2=3+ddo2133XRn2Ag@mail.gmail.com>

Hi All,

Does Kamailio has a mandatory requirement for SIP Invite to have Route
header? Although RFC-3261 doesn't have.

My requirement is to take out route header from the client if I do that I
don't see PRACK being send out from Kamilio to Media GW which causes call
drop eventually in 15sec. Any thoughts..........

BR, Apoorv
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From govoiper at gmail.com  Thu Aug 16 10:29:35 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 16 Aug 2012 13:29:35 +0500
Subject: [SR-Users] rtpproxy module issue
In-Reply-To: <CAEprJzL2-fLfqgaLYfO5A9tBsbDPzb1WtG7-2qqHCmMUL0zzVA@mail.gmail.com>
References: <CAEprJzL2-fLfqgaLYfO5A9tBsbDPzb1WtG7-2qqHCmMUL0zzVA@mail.gmail.com>
Message-ID: <CAJUJwtjB7fmm1pa60wJMLPPte=Qu_AeZ1z2aqysQaZch_Pb+dQ@mail.gmail.com>

Hi,

Though not an expert, I got few things in mind for your case. In the
upgrade process the modules folder remain the same ? right ! if that is so
is there any chances that the new installation skipped installation of
newer rtpproxy module and hence the old version rtpproxy module could've
been left in place. That could be a reason for not identifying the older
module.
This whole speculation could be wrong if the newer version supports the
older version modules.

BR
Sammy


On Thu, Aug 16, 2012 at 10:56 AM, phillman25 <phillman25 at gmail.com> wrote:

> Dear List
>
> I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1
> However, when trying to start kamailio its giving me the below error:
>
>
> ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module
> </usr/local/lib/kamailio/modules_k/rtpproxy.so>:
> /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc
> : <core> [cfg.y:3591]: parse error in config file
> /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
> load module
>
>
> Does anyone know why rtpproxy module cannot be loaded?
>
>
> This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so:
>
>
>
> shell-init: error retrieving current directory: getcwd: cannot access
> parent directories: No such file or directory
>         linux-gate.so.1 =>  (0x00276000)
>         libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000)
>         libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000)
>         libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000)
>         /lib/ld-linux.so.2 (0x00114000)
>
>
>
> Regards Phillip
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From klaus.mailinglists at pernau.at  Thu Aug 16 11:45:57 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Thu, 16 Aug 2012 11:45:57 +0200
Subject: [SR-Users] Is Route Header in SIP Invite Mandatory?
In-Reply-To: <CAOfMT3O8bONUwq1HTzy92_WC89042zUmx2=3+ddo2133XRn2Ag@mail.gmail.com>
References: <CAOfMT3O8bONUwq1HTzy92_WC89042zUmx2=3+ddo2133XRn2Ag@mail.gmail.com>
Message-ID: <502CC155.5030603@pernau.at>

On 16.08.2012 09:40, Apoorv Khare wrote:
> Hi All,
>
> Does Kamailio has a mandatory requirement for SIP Invite to have Route
> header? Although RFC-3261 doesn't have.

Kamailio does not have a requirement on Route headers. Although you can 
check manually for a Router header and handle requests different.

> My requirement is to take out route header from the client if I do that
> I don't see PRACK being send out from Kamilio to Media GW which causes
> call drop eventually in 15sec. Any thoughts..........

Maybe you can re-phrase your problem. I didn't understand it.

regards
Klaus


From phillman25 at gmail.com  Thu Aug 16 13:19:56 2012
From: phillman25 at gmail.com (phillman25)
Date: Thu, 16 Aug 2012 14:19:56 +0300
Subject: [SR-Users] rtpproxy module issue
Message-ID: <CAEprJzJKn+OFDdrpFbetFG4tyu5beh7+R0mOarYo4ueS8awkmA@mail.gmail.com>

Dear SamyGo

Thanks for your response.

Yes you're right a folder called rtpproxy.so was left in the
directory /usr/local/lib/kamailio/modules_k that had not been updated to
the date kamailio was compiled and once i deleted it everything was then
ok. I checked in directory /usr/local/lib/kamailio/modules and there i
found a new rtpproxy.so folder that was updated on the installation date.
Seems like rtpproxy.so in the new version is installed
in /usr/local/lib/kamailio/modules and not
in /usr/local/lib/kamailio/modules_k. Is that correct?

Well everything is working ok now!

Thanks for your assistance!
Regards
Phillip



> Message: 4
> Date: Thu, 16 Aug 2012 13:29:35 +0500
> From: SamyGo <govoiper at gmail.com>
> Subject: Re: [SR-Users] rtpproxy module issue
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>         Users   Mailing List" <sr-users at lists.sip-router.org>
> Message-ID:
>         <CAJUJwtjB7fmm1pa60wJMLPPte=
> Qu_AeZ1z2aqysQaZch_Pb+dQ at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> Though not an expert, I got few things in mind for your case. In the
> upgrade process the modules folder remain the same ? right ! if that is so
> is there any chances that the new installation skipped installation of
> newer rtpproxy module and hence the old version rtpproxy module could've
> been left in place. That could be a reason for not identifying the older
> module.
> This whole speculation could be wrong if the newer version supports the
> older version modules.
>
> BR
> Sammy
>
>
> On Thu, Aug 16, 2012 at 10:56 AM, phillman25 <phillman25 at gmail.com> wrote:
>
> > Dear List
> >
> > I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1
> > However, when trying to start kamailio its giving me the below error:
> >
> >
> > ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open
> module
> > </usr/local/lib/kamailio/modules_k/rtpproxy.so>:
> > /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol:
> qm_realloc
> > : <core> [cfg.y:3591]: parse error in config file
> > /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
> > load module
> >
> >
> > Does anyone know why rtpproxy module cannot be loaded?
> >
> >
> > This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so:
> >
> >
> >
> > shell-init: error retrieving current directory: getcwd: cannot access
> > parent directories: No such file or directory
> >         linux-gate.so.1 =>  (0x00276000)
> >         libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000)
> >         libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1
> (0x00768000)
> >         libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000)
> >         /lib/ld-linux.so.2 (0x00114000)
> >
> >
> >
> > Regards Phillip
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> -------------- next part --------------
> An HTML attachment was scrubbed...
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> http://lists.sip-router.org/pipermail/sr-users/attachments/20120816/919fdda5/attachment-0001.htm
> >
>
> ------------------------------
>
> Message: 5
> Date: Thu, 16 Aug 2012 11:45:57 +0200
> From: Klaus Darilion <klaus.mailinglists at pernau.at>
> Subject: Re: [SR-Users] Is Route Header in SIP Invite Mandatory?
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>         Users   Mailing List" <sr-users at lists.sip-router.org>
> Message-ID: <502CC155.5030603 at pernau.at>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 16.08.2012 09:40, Apoorv Khare wrote:
> > Hi All,
> >
> > Does Kamailio has a mandatory requirement for SIP Invite to have Route
> > header? Although RFC-3261 doesn't have.
>
> Kamailio does not have a requirement on Route headers. Although you can
> check manually for a Router header and handle requests different.
>
> > My requirement is to take out route header from the client if I do that
> > I don't see PRACK being send out from Kamilio to Media GW which causes
> > call drop eventually in 15sec. Any thoughts..........
>
> Maybe you can re-phrase your problem. I didn't understand it.
>
> regards
> Klaus
>
>
>
> ------------------------------
>
> _______________________________________________
> sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> End of sr-users Digest, Vol 87, Issue 52
> ****************************************
>
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From carlos.ruizdiaz at gmail.com  Thu Aug 16 17:25:45 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Thu, 16 Aug 2012 11:25:45 -0400
Subject: [SR-Users] append_branch with pseudo variables
Message-ID: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>

Hello guys,

Why I can only append_branch() a SIP URI as a literal constant string? Why
not from a pseudovariable? It's the same for most of the core functions.

My intention is to do something like this:

append_branch("$cnx3a(route1)");
t_relay();

where cnx3a is a variable containing the SIP URIs.

The only "fix" I have found for this is by doing it as follows:

$ru = $cnx3a(route1);
append_branch();
t_relay();

Is there any other (and more elegant) way?

Regards.

Carlos.
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From miconda at gmail.com  Thu Aug 16 18:13:01 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 16 Aug 2012 18:13:01 +0200
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
Message-ID: <502D1C0D.7050901@gmail.com>

Hello,


On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
> Hello guys,
>
> Why I can only append_branch() a SIP URI as a literal constant string? 
> Why not from a pseudovariable? It's the same for most of the core 
> functions.
>
> My intention is to do something like this:
>
> append_branch("$cnx3a(route1)");
> t_relay();
>
> where cnx3a is a variable containing the SIP URIs.
>
> The only "fix" I have found for this is by doing it as follows:
>
> $ru = $cnx3a(route1);
> append_branch();
> t_relay();
>
> Is there any other (and more elegant) way?

you can use km_append_branch:

http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From miconda at gmail.com  Thu Aug 16 18:14:46 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 16 Aug 2012 18:14:46 +0200
Subject: [SR-Users] Suggestions for time utils
In-Reply-To: <CACr1DbDYVaVkUys41aDo+1mK+vDxP9TQJ6LRH1eVzfQaGAy6Fg@mail.gmail.com>
References: <4EEF3500.6060800@sipwise.com> <4EEF8272.9090209@gmail.com>
	<4EEF8792.9040304@sipwise.com> <4EEF96B4.2050806@gmail.com>
	<CACr1DbAyjd4+-0AYVfXQKAg7FAVpZQOzLtvOTPigGx9Yvigrwg@mail.gmail.com>
	<502A8A6E.906@gmail.com>
	<69D31FDB-B6E7-4476-8014-D0B1CAAC1B39@cryy.com>
	<502AA13F.4030203@gmail.com>
	<CACr1DbDyx3pg13YVTrGpeyZY0K0f1P4xP+Ui=nEjUpd8FHw_SA@mail.gmail.com>
	<502AAB71.5060508@gmail.com>
	<CACr1DbCf0W2O2=WK8pt0qL7EpNsAnCySh_KEJfwUdtkjkBPfBA@mail.gmail.com>
	<502AB0E5.10303@gmail.com>
	<CACr1DbDt2N5SAYLZF40GF7FAHDAzc_yLVbCP+f=SvEdWm1955w@mail.gmail.com>
	<502AB36D.7080006@gmail.com>
	<CACr1DbDYVaVkUys41aDo+1mK+vDxP9TQJ6LRH1eVzfQaGAy6Fg@mail.gmail.com>
Message-ID: <502D1C76.9050203@gmail.com>

Hello,

On 8/14/12 11:14 PM, Brandon Armstead wrote:
> Daniel,
>
>    One question I do have - is the $timef(%Y%m%d)  evaluated only once 
> at run time (when kamailio is started) - or is it evaluated at the 
> time of the ACC'ing action.

it is evaluated at runtime.

Cheers,
Daniel

>
> Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 1:22 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     the $timef(...) is evaluated, but you don't have time specifiers
>     there, only static letters. You have to use % in front of the
>     letters, as I guess from the example:
>
>     http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables
>
>     Try acc_$timef(%Y%m%d).
>
>     Cheers,
>     Daniel
>
>
>
>     On 8/14/12 10:17 PM, Brandon Armstead wrote:
>>     Daniel,
>>
>>         My apologies - forgot to make clean.  However still no cigar.
>>
>>     Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
>>     [km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd'
>>     doesn't exist
>>     Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc
>>     [acc.c:405]: failed to insert into database
>>
>>     Sincerely,
>>     Brandon Armstead
>>
>>     On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         are you sure you re-installed and the right acc.so is used?
>>
>>         The support is for generic PV, if it works with one it should
>>         work with any. Which one does work for you?
>>
>>         Can you give exact table name as printed in the SQL query?
>>         Does it have the parenthesis and the format string?
>>
>>         Cheers,
>>         Daniel
>>
>>
>>         On 8/14/12 9:55 PM, Brandon Armstead wrote:
>>>         Daniel,
>>>
>>>            Patch applies fine - it still does not seem to take when
>>>         using $timef, i..e
>>>
>>>         modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>>>
>>>         It looks as if the $timef is not being interpreted and is
>>>         simply writing to acc'ing i.e. INSERT INTO acc_$timef
>>>
>>>         Thanks!
>>>
>>>         Sincerely,
>>>         Brandon Armstead
>>>         On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla
>>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>
>>>             Hello,
>>>
>>>             I reapplied the patch (cherry-picked from the initial
>>>             one) to the master branch.
>>>
>>>             Can you test that and see if it works fine?
>>>
>>>             You can cherry-picked to your branch, try:
>>>
>>>             git pull origin
>>>             git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>>
>>>             If it applies fine, compiles and the test go ok, then I
>>>             will backport as soon as Juha confirms that was no solid
>>>             reason in discarding this feature by his commit.
>>>
>>>             Cheers,
>>>             Daniel
>>>
>>>
>>>             On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>>>             Daniel,
>>>>
>>>>                In my research I saw that commit as well but figured
>>>>             it was some kind of weird merging error.  Thanks for
>>>>             your time!  Look forward to hearing back from you guys.
>>>>
>>>>             Sincerely,
>>>>             Brandon Armstead
>>>>
>>>>             On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin
>>>>             Mierla <miconda at gmail.com <mailto:miconda at gmail.com>>
>>>>             wrote:
>>>>
>>>>                 Hello,
>>>>
>>>>                 checked the sources and it seems that Juha reverted
>>>>                 this feature with the commit:
>>>>
>>>>                 http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>>
>>>>                 I guess it was accidentally, I will ask on devels list.
>>>>
>>>>                 Cheers,
>>>>                 Daniel
>>>>
>>>>
>>>>
>>>>                 On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>>>                 Let me also add that I am using 3.2 but I see the
>>>>>                 commit was quite some time ago so I have a feeling
>>>>>                 it wouldn't work in 3.3 either.  Thanks!
>>>>>
>>>>>                 Sent from my iPhone
>>>>>
>>>>>                 On Aug 14, 2012, at 10:27 AM, Daniel-Constantin
>>>>>                 Mierla <miconda at gmail.com
>>>>>                 <mailto:miconda at gmail.com>> wrote:
>>>>>
>>>>>>                 Hello,
>>>>>>
>>>>>>                 this functionality should be already there.
>>>>>>                 Doesn't work for you?
>>>>>>
>>>>>>                 Cheers,
>>>>>>                 Daniel
>>>>>>
>>>>>>                 On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>>>>                 Sorry to wake up an old thread.
>>>>>>>
>>>>>>>                 However - I am looking to export this $timef
>>>>>>>                 function to the param initialization for
>>>>>>>                 accounting, i.e. db_table_acc
>>>>>>>
>>>>>>>                 modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>>>>
>>>>>>>                 I've looked into completing this myself however
>>>>>>>                 I simply am not familiar enough at this point
>>>>>>>                 between the three different modules that it
>>>>>>>                 would take to implement this (acc, dbsr1, pv).
>>>>>>>
>>>>>>>                 Look forward to any help / insight you may be
>>>>>>>                 able to provide.
>>>>>>>
>>>>>>>                 Thanks as always!
>>>>>>>
>>>>>>>                 Sincerely,
>>>>>>>                 Brandon Armstead
>>>>>>>
>>>>>>>                 On Mon, Dec 19, 2011 at 11:55 AM,
>>>>>>>                 Daniel-Constantin Mierla <miconda at gmail.com
>>>>>>>                 <mailto:miconda at gmail.com>> wrote:
>>>>>>>
>>>>>>>                     Hello,
>>>>>>>
>>>>>>>
>>>>>>>                     On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>>>>
>>>>>>>                         Hi Daniel,
>>>>>>>
>>>>>>>                         On 12/19/2011 07:29 PM,
>>>>>>>                         Daniel-Constantin Mierla wrote:
>>>>>>>
>>>>>>>                             I don't know what are all the
>>>>>>>                             functions you think of, but for the
>>>>>>>                             example provided above, config file
>>>>>>>                             does it easy right now. There is a
>>>>>>>                             pseudo-variable that gives
>>>>>>>                             broken-time attribute that can be
>>>>>>>                             used with
>>>>>>>                             avp_check(), iirc, should be:
>>>>>>>
>>>>>>>                             avp_db_load(...);
>>>>>>>                             if(avp_check("$time(wday)",
>>>>>>>                             "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>>>
>>>>>>>                             Of course there is the option of
>>>>>>>                             doing while loop, but maybe gets to
>>>>>>>                             large for desired config file.
>>>>>>>
>>>>>>>                         This is pretty much what I had in mind
>>>>>>>                         with my new functions/module, but
>>>>>>>                         I've completely overlooked that PV when
>>>>>>>                         searching the docs for this
>>>>>>>                         feature. Thank you very much for
>>>>>>>                         pointing that out!
>>>>>>>
>>>>>>>                     for sake of public knowledge, just to add on
>>>>>>>                     time specific features: there is also
>>>>>>>                     $timef(format) which returns current time
>>>>>>>                     attributes based on strftime specifiers --
>>>>>>>                     its documentation was missing, I just added
>>>>>>>                     it. Also, there is a transformation
>>>>>>>                     {s.ftime,format) which can take any integer
>>>>>>>                     variable holding timestamp and return value
>>>>>>>                     based on strftime format.
>>>>>>>
>>>>>>>
>>>>>>>                     Cheers,
>>>>>>>                     Daniel
>>>>>>>
>>>>>>>                     -- 
>>>>>>>                     Daniel-Constantin Mierla --
>>>>>>>                     http://www.asipto.com
>>>>>>>                     http://linkedin.com/in/miconda --
>>>>>>>                     http://twitter.com/miconda
>>>>>>>
>>>>>>>
>>>>>>>                     _______________________________________________
>>>>>>>                     SIP Express Router (SER) and Kamailio
>>>>>>>                     (OpenSER) - sr-users mailing list
>>>>>>>                     sr-users at lists.sip-router.org
>>>>>>>                     <mailto:sr-users at lists.sip-router.org>
>>>>>>>                     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>                 -- 
>>>>>>                 Daniel-Constantin Mierla -http://www.asipto.com
>>>>>>                 http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>>>>                 Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>>>>                 Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>>>
>>>>                 -- 
>>>>                 Daniel-Constantin Mierla -http://www.asipto.com
>>>>                 http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>>                 Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>>                 Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>>>
>>>>
>>>
>>>             -- 
>>>             Daniel-Constantin Mierla -http://www.asipto.com
>>>             http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>>             Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>>             Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>>
>>>
>>
>>         -- 
>>         Daniel-Constantin Mierla -http://www.asipto.com
>>         http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>         Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>>         Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>     Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From miconda at gmail.com  Thu Aug 16 18:17:30 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 16 Aug 2012 18:17:30 +0200
Subject: [SR-Users] sms module & IMS
In-Reply-To: <CAGrTonSCfYEmMPqvLvwx9GWQ=eKE1WADbxfO1D0mhZaScoBrLQ@mail.gmail.com>
References: <CAGrTonSCfYEmMPqvLvwx9GWQ=eKE1WADbxfO1D0mhZaScoBrLQ@mail.gmail.com>
Message-ID: <502D1D1A.1070402@gmail.com>

Hello,

the module is working with phones having an AT modem for sending SMS.

You can set debug=3 in the config file and send the output here to 
investigate, in case you don't understand exactly what happens.

Cheers,
Daniel

On 8/14/12 5:10 PM, Oleksandr Shchirskyi wrote:
> Hi,
>
> I'm using OpenIMSCore based on the SER. At the moment I try to add
> SMS-GW to it.
> I've found, that SER has sms module, and I tried to use it. But,
> unfortunately, I couldn't make it work.
>
> It's not very clear for me how SIP MESSAGEs can be routed to the sms
> module. This module works not as application server (or I'm wrong?)
>
> At the moment I have such configuration in the scscf.cfg:
>
> ==============================
> modparam("sms","modems","Nokia [d=/dev/ttyACM0;b=9600]")
> modparam("sms","networks","D1[m=10]")
> modparam("sms","links","Nokia[D1]")
> modparam("sms","domain","open-ims.test")
> modparam("sms","max_sms_parts",3)
> modparam("sms","use_contact",1)
>
> # main routing logic
>
> route{
>          if(method=="MESSAGE"){
>                  sms_send_msg_to_net("D1");
>          }
> ...
> }
>
> ==============================
>
> As a hardware GSM gateway my cell phone is using.
> As I see, nothing happens when s-cscf processes SIP MESSAGEs. I don't
> see that these messages are forwarded to the sms module, or smth like
> that.
>
> Final question - is it possible to use this module as IP-SMS-GW for
> the OpenIMSCore? And how?
>
> I would be very appreciated for any help.
> Thank you in advance.
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From simon at ganesh-consulting.ch  Thu Aug 16 14:10:12 2012
From: simon at ganesh-consulting.ch (Simon Hintermann)
Date: Thu, 16 Aug 2012 14:10:12 +0200
Subject: [SR-Users] ADSL client-side problem
Message-ID: <4501cdc1-a894-4b5d-837c-7f68c136cbcd@ganesh-consulting.ch>

Hello,
 
I am experiencing some disconnections with my VOIP phones.
 
Let me explain:
 
My server is an Asterisk + Kamailio in a datacenter with a fixed IP, and a bridge firewall in front of it, so no NAT here.
 
My client (several offices) have usually no problem when they have a fixed IP, but encounter fatal disconnection when they have a dynamic IP. I had the case with Siemens, SNOM and Aastra phones, without any difference. They are of course natted behind an ADSL router. I also have clients doing load-balance between two ADSL lines, which is also problematic.
 
The only way for me to recover is to change the private IP of the VOIP phones. Seems more like a TCP problem or a routing problem, but I am no network guru.
 
I have two ADSL lines to test it, and when I switch from one line to another, every phone becomes "Not registered". Even falling back tot he first line does not help recovering. I did not try to wait for longer than 24 hours to recover.
 
So my question is, more generally: is it possible to allow the clients (VOIP phones) to re-connect with a new IP address without having closed correctly the first connection (SIP session, I presume)? It seems not by default, but I played with the timeouts server-side and phone-side, without luck.
 
I saw many posts with PBX having problems when behind a dynamic IP, but nothing about the clients having dynamic addresses or load-balancing. Normally, having a dynamic IP does not mean that your IP will change << on-the-fly >>, but each time you shut down and reboot later your ADSL router, but we came to the conclusion that every client NOT having a fixed IP was experiencing fatal disconnects. So I managed to test and validate this behavior with our two ADSL lines.
 
Hope someone can help me.
 
Greetings
 
Simon
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From carlos.ruizdiaz at gmail.com  Thu Aug 16 20:17:02 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Thu, 16 Aug 2012 14:17:02 -0400
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <502D1C0D.7050901@gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
Message-ID: <CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>

Great! I'll give it a try.

Thanks!

Carlos.

On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

> Hello,
>
>
>
> On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>
>> Hello guys,
>>
>> Why I can only append_branch() a SIP URI as a literal constant string?
>> Why not from a pseudovariable? It's the same for most of the core functions.
>>
>> My intention is to do something like this:
>>
>> append_branch("$cnx3a(route1)"**);
>> t_relay();
>>
>> where cnx3a is a variable containing the SIP URIs.
>>
>> The only "fix" I have found for this is by doing it as follows:
>>
>> $ru = $cnx3a(route1);
>> append_branch();
>> t_relay();
>>
>> Is there any other (and more elegant) way?
>>
>
> you can use km_append_branch:
>
> http://kamailio.org/docs/**modules/stable/modules_k/kex.**html#id2551404<http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404>
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda>
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
> http://asipto.com/u/kpw
>
>
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From carlos.ruizdiaz at gmail.com  Thu Aug 16 21:15:49 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Thu, 16 Aug 2012 15:15:49 -0400
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
Message-ID: <CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>

Looks like km_append_branch() doesn't have all of the signatures that the
original append_branch() has. I can't add q values for serial/parallel
forking with km_append_branch().

Also, apparently there's no way of setting the RURI by calling a function
that does that. I guess I'll have to do it using assignation on $ru
pseudovariable and writing a route to manage the serial forking I need.

Thanks.

Carlos.

On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com
> wrote:

> Great! I'll give it a try.
>
> Thanks!
>
> Carlos.
>
>
> On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>> Hello,
>>
>>
>>
>> On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>>
>>> Hello guys,
>>>
>>> Why I can only append_branch() a SIP URI as a literal constant string?
>>> Why not from a pseudovariable? It's the same for most of the core functions.
>>>
>>> My intention is to do something like this:
>>>
>>> append_branch("$cnx3a(route1)"**);
>>> t_relay();
>>>
>>> where cnx3a is a variable containing the SIP URIs.
>>>
>>> The only "fix" I have found for this is by doing it as follows:
>>>
>>> $ru = $cnx3a(route1);
>>> append_branch();
>>> t_relay();
>>>
>>> Is there any other (and more elegant) way?
>>>
>>
>> you can use km_append_branch:
>>
>> http://kamailio.org/docs/**modules/stable/modules_k/kex.**html#id2551404<http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404>
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda>
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>> http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>> http://asipto.com/u/kpw
>>
>>
>
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From miconda at gmail.com  Thu Aug 16 22:30:02 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 16 Aug 2012 22:30:02 +0200
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
	<CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
Message-ID: <502D584A.8070104@gmail.com>


On 8/16/12 9:15 PM, Carlos Ruiz D?az wrote:
> Looks like km_append_branch() doesn't have all of the signatures that 
> the original append_branch() has. I can't add q values for 
> serial/parallel forking with km_append_branch().
What signature are you looking for?

>
> Also, apparently there's no way of setting the RURI by calling a 
> function that does that.

What do you mean here? I don't understand.

Cheers,
Daniel

> I guess I'll have to do it using assignation on $ru pseudovariable and 
> writing a route to manage the serial forking I need.
>
> Thanks.
>
> Carlos.
>
> On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az 
> <carlos.ruizdiaz at gmail.com <mailto:carlos.ruizdiaz at gmail.com>> wrote:
>
>     Great! I'll give it a try.
>
>     Thanks!
>
>     Carlos.
>
>
>     On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla
>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>         Hello,
>
>
>
>         On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>
>             Hello guys,
>
>             Why I can only append_branch() a SIP URI as a literal
>             constant string? Why not from a pseudovariable? It's the
>             same for most of the core functions.
>
>             My intention is to do something like this:
>
>             append_branch("$cnx3a(route1)");
>             t_relay();
>
>             where cnx3a is a variable containing the SIP URIs.
>
>             The only "fix" I have found for this is by doing it as
>             follows:
>
>             $ru = $cnx3a(route1);
>             append_branch();
>             t_relay();
>
>             Is there any other (and more elegant) way?
>
>
>         you can use km_append_branch:
>
>         http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404
>
>         Cheers,
>         Daniel
>
>         -- 
>         Daniel-Constantin Mierla - http://www.asipto.com
>         http://twitter.com/#!/miconda
>         <http://twitter.com/#%21/miconda> -
>         http://www.linkedin.com/in/miconda
>         Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>         http://asipto.com/u/katu
>         Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>         http://asipto.com/u/kpw
>
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From carlos.ruizdiaz at gmail.com  Thu Aug 16 22:50:43 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Thu, 16 Aug 2012 16:50:43 -0400
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <502D584A.8070104@gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
	<CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
	<502D584A.8070104@gmail.com>
Message-ID: <CA+3gBcn7V4od7=ATHZUC7zgNwFvQSuZKy2vMyNjGE5fq5LotWg@mail.gmail.com>

On Thu, Aug 16, 2012 at 4:30 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>
> On 8/16/12 9:15 PM, Carlos Ruiz D?az wrote:
>
> Looks like km_append_branch() doesn't have all of the signatures that the
> original append_branch() has. I can't add q values for serial/parallel
> forking with km_append_branch().
>
> What signature are you looking for?
>

The latter is the one I'm looking for, but with the possibility of using
pseudovars.

append_branch() overloads:

- append_branch();
- append_branch("sip:a at abc.com");
- append_branch("sip:a at abc.com", "0.1");


>
>
>  Also, apparently there's no way of setting the RURI by calling a function
> that does that.
>
>
> What do you mean here? I don't understand.
>

I'm was talking about setting the request-uri from a PV.

For example: seturi("$var(my_uri)");

But it works using assignation.

$ru = $var(my_uri);



>
> Cheers,
> Daniel
>
>
>  I guess I'll have to do it using assignation on $ru pseudovariable and
> writing a route to manage the serial forking I need.
>
>  Thanks.
>
>  Carlos.
>
> On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az <
> carlos.ruizdiaz at gmail.com> wrote:
>
>> Great! I'll give it a try.
>>
>>  Thanks!
>>
>>  Carlos.
>>
>>
>> On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>>
>>> On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>>>
>>>> Hello guys,
>>>>
>>>> Why I can only append_branch() a SIP URI as a literal constant string?
>>>> Why not from a pseudovariable? It's the same for most of the core functions.
>>>>
>>>> My intention is to do something like this:
>>>>
>>>> append_branch("$cnx3a(route1)");
>>>> t_relay();
>>>>
>>>> where cnx3a is a variable containing the SIP URIs.
>>>>
>>>> The only "fix" I have found for this is by doing it as follows:
>>>>
>>>> $ru = $cnx3a(route1);
>>>> append_branch();
>>>> t_relay();
>>>>
>>>> Is there any other (and more elegant) way?
>>>>
>>>
>>>  you can use km_append_branch:
>>>
>>> http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.com
>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>>> http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>>> http://asipto.com/u/kpw
>>>
>>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From spencer at 5ninesolutions.com  Thu Aug 16 22:59:48 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 16 Aug 2012 13:59:48 -0700
Subject: [SR-Users] Kamailio NAT traversal
Message-ID: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>

Hello,
I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal.  Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints?  

My thought is something like this:

natted endpoint -> kamailio -> freeswitch
If the response is 200OK, save the registration.

Thanks,
Spencer



From danb.lists at gmail.com  Fri Aug 17 10:47:21 2012
From: danb.lists at gmail.com (DanB)
Date: Fri, 17 Aug 2012 10:47:21 +0200
Subject: [SR-Users] Accessing received after lookup()
In-Reply-To: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
Message-ID: <502E0519.9020906@gmail.com>

Hey All,

I was wondering if anyone could point me to the right direction on the 
following issue:

I use registrar with path to store contacts after main proxy. There I 
have properly stored received db field (original address of the NAT 
router).
When I query the registrar, I get the user contact details (natted ones) 
in 302 redirect. However at this point I cannot access received avp (or 
I did not find a way yet, except calling reg_fetch_contacts which I 
would prefer not to since it doubles the work lookup does).

Is there any catch which could save me in this scenario? After loopkup() 
is there any pseudovariable which I can inspect to find out user's 
contact data? Any other way to extract received contact data out of 
contact info?

Thanks in advance for any kind of tip!

DanB


From danb.lists at gmail.com  Fri Aug 17 11:13:46 2012
From: danb.lists at gmail.com (DanB)
Date: Fri, 17 Aug 2012 11:13:46 +0200
Subject: [SR-Users] Accessing received after lookup()
In-Reply-To: <502E0519.9020906@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502E0519.9020906@gmail.com>
Message-ID: <502E0B4A.8090201@gmail.com>

If it helps someone else, got in the mean time another way of accessing 
it, via transformations on top of $du:

$(du{uri.param,received}).

The question which still stands: is there more "automated" way to 
properly handle 302 redirect when the contact is behind NAT?

DanB

O



From miconda at gmail.com  Fri Aug 17 11:30:34 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 17 Aug 2012 11:30:34 +0200
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <CA+3gBcn7V4od7=ATHZUC7zgNwFvQSuZKy2vMyNjGE5fq5LotWg@mail.gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
	<CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
	<502D584A.8070104@gmail.com>
	<CA+3gBcn7V4od7=ATHZUC7zgNwFvQSuZKy2vMyNjGE5fq5LotWg@mail.gmail.com>
Message-ID: <502E0F3A.2040401@gmail.com>


On 8/16/12 10:50 PM, Carlos Ruiz D?az wrote:
>
>
> On Thu, Aug 16, 2012 at 4:30 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>
>     On 8/16/12 9:15 PM, Carlos Ruiz D?az wrote:
>>     Looks like km_append_branch() doesn't have all of the signatures
>>     that the original append_branch() has. I can't add q values for
>>     serial/parallel forking with km_append_branch().
>     What signature are you looking for?
>
>
> The latter is the one I'm looking for, but with the possibility of 
> using pseudovars.
>
> append_branch() overloads:
>
> - append_branch();
> - append_branch("sip:a at abc.com <mailto:sip%3Aa at abc.com>");
> - append_branch("sip:a at abc.com <mailto:sip%3Aa at abc.com>", "0.1");

I am considering adding the second parameter to km_append_branch() for 
backward compatibility reasons.

Alternative is to use $branch(...) variable to set the q after you 
called append branch:

http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#branch_attributes

Q has to be set via integer value, with range from 0 to 100 (0.1 would 
be 10).

>
>>
>> Also, apparently there's no way of setting the RURI by calling a 
>> function that does that.
>
>     What do you mean here? I don't understand.
>
>
> I'm was talking about setting the request-uri from a PV.
>
> For example: seturi("$var(my_uri)");
>
> But it works using assignation.
>
> $ru = $var(my_uri);

The core functions were added before the existence of cfg variables, 
they take only static strings as parameters. The way to do it is via 
assignments.

Cheers,
Daniel

>
>
>     Cheers,
>     Daniel
>
>
>>     I guess I'll have to do it using assignation on $ru
>>     pseudovariable and writing a route to manage the serial forking I
>>     need.
>>
>>     Thanks.
>>
>>     Carlos.
>>
>>     On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az
>>     <carlos.ruizdiaz at gmail.com <mailto:carlos.ruizdiaz at gmail.com>> wrote:
>>
>>         Great! I'll give it a try.
>>
>>         Thanks!
>>
>>         Carlos.
>>
>>
>>         On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla
>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>             Hello,
>>
>>
>>
>>             On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>>
>>                 Hello guys,
>>
>>                 Why I can only append_branch() a SIP URI as a literal
>>                 constant string? Why not from a pseudovariable? It's
>>                 the same for most of the core functions.
>>
>>                 My intention is to do something like this:
>>
>>                 append_branch("$cnx3a(route1)");
>>                 t_relay();
>>
>>                 where cnx3a is a variable containing the SIP URIs.
>>
>>                 The only "fix" I have found for this is by doing it
>>                 as follows:
>>
>>                 $ru = $cnx3a(route1);
>>                 append_branch();
>>                 t_relay();
>>
>>                 Is there any other (and more elegant) way?
>>
>>
>>             you can use km_append_branch:
>>
>>             http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404
>>
>>             Cheers,
>>             Daniel
>>
>>             -- 
>>             Daniel-Constantin Mierla - http://www.asipto.com
>>             http://twitter.com/#!/miconda
>>             <http://twitter.com/#%21/miconda> -
>>             http://www.linkedin.com/in/miconda
>>             Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012
>>             - http://asipto.com/u/katu
>>             Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012
>>             - http://asipto.com/u/kpw
>>
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>     Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From miconda at gmail.com  Fri Aug 17 11:37:00 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 17 Aug 2012 11:37:00 +0200
Subject: [SR-Users] Accessing received after lookup()
In-Reply-To: <502E0519.9020906@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502E0519.9020906@gmail.com>
Message-ID: <502E10BC.3040602@gmail.com>

Hello,

On 8/17/12 10:47 AM, DanB wrote:
> Hey All,
>
> I was wondering if anyone could point me to the right direction on the 
> following issue:
>
> I use registrar with path to store contacts after main proxy. There I 
> have properly stored received db field (original address of the NAT 
> router).

did you take the received parameter from the Path header? Otherwise 
received (ie. the received column value) is usually the source ip/port. 
The received parameter of Path is stored along with it in the 
appropriate column.

Cheers,
Daniel

> When I query the registrar, I get the user contact details (natted 
> ones) in 302 redirect. However at this point I cannot access received 
> avp (or I did not find a way yet, except calling reg_fetch_contacts 
> which I would prefer not to since it doubles the work lookup does).
>
> Is there any catch which could save me in this scenario? After 
> loopkup() is there any pseudovariable which I can inspect to find out 
> user's contact data? Any other way to extract received contact data 
> out of contact info?
>
> Thanks in advance for any kind of tip!
>
> DanB
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From danb.lists at gmail.com  Fri Aug 17 11:41:45 2012
From: danb.lists at gmail.com (DanB)
Date: Fri, 17 Aug 2012 11:41:45 +0200
Subject: [SR-Users] Accessing received after lookup()
In-Reply-To: <502E10BC.3040602@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502E0519.9020906@gmail.com> <502E10BC.3040602@gmail.com>
Message-ID: <502E11D9.4080703@gmail.com>

Hey Daniel,

Yeah, got the header out of received from path. Actually I have 
de-coupled registrations which work and store just fine the received 
parameter out of INVITEs which are querying the registrar using lookup. 
In the end I have ended up using received parameter if present in $du 
after lookup and appending it to contact (removing it after in the proxy 
which sits in front of registrar). Here is the code snippet for anyone 
needing it in the future:

"""
	if (!lookup("location")) {
		$var(rc) = $rc;
		t_newtran();
		switch ($var(rc)) {
			case -1:
			case -3:
				send_reply("404", "Not Found");
				exit;
			case -2:
				send_reply("405", "Method Not Allowed");
				exit;
		}
	}
	# Populate received contact parameter if found in db
	$var(received) = $(du{uri.param,received});
	if !strempty( $var(received) ) {
		$ru = $ru + ";rcv="+ $var(received);
	}
	sl_send_reply("302", "Locator Redirect");
	exit;
"""

Thanks,
DanB

On 08/17/2012 11:37 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 8/17/12 10:47 AM, DanB wrote:
>> Hey All,
>>
>> I was wondering if anyone could point me to the right direction on the
>> following issue:
>>
>> I use registrar with path to store contacts after main proxy. There I
>> have properly stored received db field (original address of the NAT
>> router).
>
> did you take the received parameter from the Path header? Otherwise
> received (ie. the received column value) is usually the source ip/port.
> The received parameter of Path is stored along with it in the
> appropriate column.
>
> Cheers,
> Daniel
>
>> When I query the registrar, I get the user contact details (natted
>> ones) in 302 redirect. However at this point I cannot access received
>> avp (or I did not find a way yet, except calling reg_fetch_contacts
>> which I would prefer not to since it doubles the work lookup does).
>>
>> Is there any catch which could save me in this scenario? After
>> loopkup() is there any pseudovariable which I can inspect to find out
>> user's contact data? Any other way to extract received contact data
>> out of contact info?
>>
>> Thanks in advance for any kind of tip!
>>
>> DanB
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



From miconda at gmail.com  Fri Aug 17 11:42:46 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 17 Aug 2012 11:42:46 +0200
Subject: [SR-Users] Accessing received after lookup()
In-Reply-To: <502E0B4A.8090201@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502E0519.9020906@gmail.com> <502E0B4A.8090201@gmail.com>
Message-ID: <502E1216.3040007@gmail.com>

Hello,

On 8/17/12 11:13 AM, DanB wrote:
> If it helps someone else, got in the mean time another way of 
> accessing it, via transformations on top of $du:
>
> $(du{uri.param,received}).

$du is the address of next hop and it is set due to the Path headers in 
this case. In the case there is no intermediary proxy, it will be the 
address of the NAT router.

In case you have many contacts for same users, then you have to look 
also at the other branches, which are accessible via $branch(...).

>
> The question which still stands: is there more "automated" way to 
> properly handle 302 redirect when the contact is behind NAT?
This might not work for multiple contacts. I even wonder if there is a 
RFC saying what to do in the case there are many contacts with different 
Path headers. Path headers have to be turned into Route headers after 
lookup(), but for redirects?!? Shall Path should be added in reply? Any 
IETF guru around that can shed some light?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From danb.lists at gmail.com  Fri Aug 17 12:50:19 2012
From: danb.lists at gmail.com (DanB)
Date: Fri, 17 Aug 2012 12:50:19 +0200
Subject: [SR-Users] Accessing received after lookup()
In-Reply-To: <502E1216.3040007@gmail.com>
References: <mailman.3.1344938402.10364.sr-users@lists.sip-router.org>
	<502E0519.9020906@gmail.com> <502E0B4A.8090201@gmail.com>
	<502E1216.3040007@gmail.com>
Message-ID: <502E21EB.2020303@gmail.com>

Daniel,

As always experience counts the most (I did not consider the multiple 
contacts scenario).

Here is what I cooked now after your advice (for the sake of proper 
archiving). In the end I will process the rcv contact parameter (which 
will become ruri param) in the proxy sitting in front of registrar and 
set the $du out of it.

"""
	if (!lookup("location")) {
		$var(rc) = $rc;
		t_newtran();
		switch ($var(rc)) {
			case -1:
			case -3:
				send_reply("404", "Not Found");
				exit;
			case -2:
				send_reply("405", "Method Not Allowed");
				exit;
		}
	}

	# Populate received contact parameter if found in db
	$var(received) = $(du{uri.param,received});
	if !strempty( $var(received) ) {
		$ru = $ru + ";rcv="+ $var(received);
	}
	$var(i)=0;
	while($var(i)<$branch(count)) {
		$var(received) = $(branch(dst_uri)[$var(i)]{uri.param,received});
		if !strempty( $var(received) ) {
			$(branch(uri)[$var(i)]) = $(branch(uri)[$var(i)]) + ";rcv="+ 
$var(received);
		}
    		$var(i) = $var(i) + 1;
	}

	sl_send_reply("302", "Locator Redirect");

	exit;
"""

Thanks again!

DanB

On 08/17/2012 11:42 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 8/17/12 11:13 AM, DanB wrote:
>> If it helps someone else, got in the mean time another way of
>> accessing it, via transformations on top of $du:
>>
>> $(du{uri.param,received}).
>
> $du is the address of next hop and it is set due to the Path headers in
> this case. In the case there is no intermediary proxy, it will be the
> address of the NAT router.
>
> In case you have many contacts for same users, then you have to look
> also at the other branches, which are accessible via $branch(...).
>
>>
>> The question which still stands: is there more "automated" way to
>> properly handle 302 redirect when the contact is behind NAT?
> This might not work for multiple contacts. I even wonder if there is a
> RFC saying what to do in the case there are many contacts with different
> Path headers. Path headers have to be turned into Route headers after
> lookup(), but for redirects?!? Shall Path should be added in reply? Any
> IETF guru around that can shed some light?
>
> Cheers,
> Daniel
>



From jmr.richardson at gmail.com  Fri Aug 17 14:42:46 2012
From: jmr.richardson at gmail.com (JR Richardson)
Date: Fri, 17 Aug 2012 07:42:46 -0500
Subject: [SR-Users] Request for advice,
	running SIP-Router in OpenVZ Container
Message-ID: <001201cd7c75$cde02890$69a079b0$@gmail.com>

Hi All,

 

I'm considering running Kamailio as a virtual machine, with such low
utilization, it doesn't seem to make sense to keep running it on a physical
host server.

 

I've been virtualizing Asterisk PBX's for years and run a host of other
virtualized servers with OpenVZ, VMware and MS V-Server without issues.

 

I'd like any opinions or thoughts about pitfalls or caveats I should be
aware of specific to virtualized SIP-Router/Kamailio.

 

Thanks in advanced.

 

JR

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From bh at udev.org  Fri Aug 17 15:08:18 2012
From: bh at udev.org (Benjamin Henrion)
Date: Fri, 17 Aug 2012 15:08:18 +0200
Subject: [SR-Users] Request for advice,
	running SIP-Router in OpenVZ Container
In-Reply-To: <001201cd7c75$cde02890$69a079b0$@gmail.com>
References: <001201cd7c75$cde02890$69a079b0$@gmail.com>
Message-ID: <CANjd3ndmNvNi4MLH5K0s3vStbWeo=HdPaqMxVmfrbiCnfM_eGw@mail.gmail.com>

On Friday, August 17, 2012, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
>
>
> I?m considering running Kamailio as a virtual machine, with such low
utilization, it doesn?t seem to make sense to keep running it on a physical
host server.
>
>
>
> I?ve been virtualizing Asterisk PBX?s for years and run a host of other
virtualized servers with OpenVZ, VMware and MS V-Server without issues.
>
>
>
> I?d like any opinions or thoughts about pitfalls or caveats I should be
aware of specific to virtualized SIP-Router/Kamailio.
>
>
>
> Thanks in advanced.

I run kamailios in openvz containers for 2 years now, the only problems i
encountered was with memory allocation, but with the latest vzswap
functionalities, this is fixed.

You might want to play with the mem argument that you see with ps aux so
see if you hit the limit of the openvz container.

Best,

>
>
>
> JR

-- 
Benjamin Henrion <bhenrion at ffii.org>
FFII Brussels - +32-484-566109 - +32-2-3500762
"In July 2005, after several failed attempts to legalise software patents
in Europe, the patent establishment changed its strategy. Instead of
explicitly seeking to sanction the patentability of software, they are now
seeking to create a central European patent court, which would establish
and enforce patentability rules in their favor, without any possibility of
correction by competing courts or democratically elected legislators."
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From tayeb at vobradio.org  Fri Aug 17 16:58:59 2012
From: tayeb at vobradio.org (Tayeb)
Date: Fri, 17 Aug 2012 15:58:59 +0100
Subject: [SR-Users] Request for advice,
	running SIP-Router in OpenVZ Container
In-Reply-To: <001201cd7c75$cde02890$69a079b0$@gmail.com>
References: <001201cd7c75$cde02890$69a079b0$@gmail.com>
Message-ID: <65BDA5C3-E516-42B6-A851-17A372D874F7@vobradio.org>

I do that
No issue so far

Envoy? de mon iPhone

Le 17 ao?t 2012 ? 13:42, "JR Richardson" <jmr.richardson at gmail.com> a ?crit :

> Hi All,
>  
> I?m considering running Kamailio as a virtual machine, with such low utilization, it doesn?t seem to make sense to keep running it on a physical host server.
>  
> I?ve been virtualizing Asterisk PBX?s for years and run a host of other virtualized servers with OpenVZ, VMware and MS V-Server without issues.
>  
> I?d like any opinions or thoughts about pitfalls or caveats I should be aware of specific to virtualized SIP-Router/Kamailio.
>  
> Thanks in advanced.
>  
> JR
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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From govoiper at gmail.com  Fri Aug 17 17:04:19 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 17 Aug 2012 20:04:19 +0500
Subject: [SR-Users] Kamailio NAT traversal
In-Reply-To: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>
References: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>
Message-ID: <CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>

Hi,

You kind of sound a little different here. Are you saying that the
REGISTRATIONs will be handled by Freeswitch but store the registration Data
in Kamailio "location" table !?

Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the
one on integrating the Asterisk Realtime with kamailio, In that articular
configuration file Kamailio Forwards/relays the Registration attempts to
the Media-Server (FreeSwitch in your case).

That will get your NAT thing handled atleast. Next thing is , if I'm right
about your requirement, saving of Authenticated users in the Kamailio
locations table and I'm not very sure about how to do this.


BR
Sammy


On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello,
> I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal.
>  Is there a way I can allow freeswitch to handle all auth but store the
> registration in usrloc as I need to send an options ping to the endpoints?
>
> My thought is something like this:
>
> natted endpoint -> kamailio -> freeswitch
> If the response is 200OK, save the registration.
>
> Thanks,
> Spencer
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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From fred at teamforrest.com  Fri Aug 17 17:21:21 2012
From: fred at teamforrest.com (Fred Posner)
Date: Fri, 17 Aug 2012 11:21:21 -0400
Subject: [SR-Users] Kamailio NAT traversal
In-Reply-To: <CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>
References: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>
	<CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>
Message-ID: <6508A16D-A415-4BF0-A2BB-BBE0A42B5EF3@teamforrest.com>

Hi Spencer,

Is Kamailio also natted? If so, you may have some issues... if not, it should work great. I run a server like this as well... very happy with it.

I gave up on kamailio/freeswitch behind nat. Well, didn't give up, just don't have the time to make it work.

With best regards,

Fred
http://qxork.com

On Aug 17, 2012, at 11:04 AM, SamyGo wrote:

> Hi,
> 
> You kind of sound a little different here. Are you saying that the REGISTRATIONs will be handled by Freeswitch but store the registration Data in Kamailio "location" table !? 
> 
> Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the one on integrating the Asterisk Realtime with kamailio, In that articular configuration file Kamailio Forwards/relays the Registration attempts to the Media-Server (FreeSwitch in your case). 
> 
> That will get your NAT thing handled atleast. Next thing is , if I'm right about your requirement, saving of Authenticated users in the Kamailio locations table and I'm not very sure about how to do this.
> 
> 
> BR
> Sammy
> 
> 
> On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello,
> I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal.  Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints?
> 
> My thought is something like this:
> 
> natted endpoint -> kamailio -> freeswitch
> If the response is 200OK, save the registration.
> 
> Thanks,
> Spencer



From carlos.ruizdiaz at gmail.com  Fri Aug 17 17:38:12 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Fri, 17 Aug 2012 11:38:12 -0400
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <502E0F3A.2040401@gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
	<CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
	<502D584A.8070104@gmail.com>
	<CA+3gBcn7V4od7=ATHZUC7zgNwFvQSuZKy2vMyNjGE5fq5LotWg@mail.gmail.com>
	<502E0F3A.2040401@gmail.com>
Message-ID: <CA+3gBckuke2dV9kQSEjb2F_MMvD=+0tJ2O3DTMuGkydsWcZ8jg@mail.gmail.com>

I'll try that :).

Thanks for your help.

Carlos.

On Fri, Aug 17, 2012 at 5:30 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>
> On 8/16/12 10:50 PM, Carlos Ruiz D?az wrote:
>
>
>
> On Thu, Aug 16, 2012 at 4:30 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>
>> On 8/16/12 9:15 PM, Carlos Ruiz D?az wrote:
>>
>> Looks like km_append_branch() doesn't have all of the signatures that the
>> original append_branch() has. I can't add q values for serial/parallel
>> forking with km_append_branch().
>>
>>  What signature are you looking for?
>>
>
>  The latter is the one I'm looking for, but with the possibility of using
> pseudovars.
>
>  append_branch() overloads:
>
>  - append_branch();
> - append_branch("sip:a at abc.com");
> - append_branch("sip:a at abc.com", "0.1");
>
>
> I am considering adding the second parameter to km_append_branch() for
> backward compatibility reasons.
>
> Alternative is to use $branch(...) variable to set the q after you called
> append branch:
>
>
> http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#branch_attributes
>
> Q has to be set via integer value, with range from 0 to 100 (0.1 would be
> 10).
>
>
>
>
>  Also, apparently there's no way of setting the RURI by calling a function
> that does that.
>
>
>   What do you mean here? I don't understand.
>>
>
>  I'm was talking about setting the request-uri from a PV.
>
>  For example: seturi("$var(my_uri)");
>
>  But it works using assignation.
>
>  $ru = $var(my_uri);
>
>
> The core functions were added before the existence of cfg variables, they
> take only static strings as parameters. The way to do it is via assignments.
>
> Cheers,
> Daniel
>
>
>
>
>
>>
>> Cheers,
>> Daniel
>>
>>
>>  I guess I'll have to do it using assignation on $ru pseudovariable and
>> writing a route to manage the serial forking I need.
>>
>>  Thanks.
>>
>>  Carlos.
>>
>> On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az <
>> carlos.ruizdiaz at gmail.com> wrote:
>>
>>> Great! I'll give it a try.
>>>
>>>  Thanks!
>>>
>>>  Carlos.
>>>
>>>
>>> On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla <
>>> miconda at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>>
>>>>
>>>> On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>>>>
>>>>> Hello guys,
>>>>>
>>>>> Why I can only append_branch() a SIP URI as a literal constant string?
>>>>> Why not from a pseudovariable? It's the same for most of the core functions.
>>>>>
>>>>> My intention is to do something like this:
>>>>>
>>>>> append_branch("$cnx3a(route1)");
>>>>> t_relay();
>>>>>
>>>>> where cnx3a is a variable containing the SIP URIs.
>>>>>
>>>>> The only "fix" I have found for this is by doing it as follows:
>>>>>
>>>>> $ru = $cnx3a(route1);
>>>>> append_branch();
>>>>> t_relay();
>>>>>
>>>>> Is there any other (and more elegant) way?
>>>>>
>>>>
>>>>  you can use km_append_branch:
>>>>
>>>> http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - http://www.asipto.com
>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>>>> http://asipto.com/u/katu
>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>>>> http://asipto.com/u/kpw
>>>>
>>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>
>
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From miconda at gmail.com  Fri Aug 17 17:57:25 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 17 Aug 2012 17:57:25 +0200
Subject: [SR-Users] Kamailio NAT traversal
In-Reply-To: <6508A16D-A415-4BF0-A2BB-BBE0A42B5EF3@teamforrest.com>
References: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>
	<CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>
	<6508A16D-A415-4BF0-A2BB-BBE0A42B5EF3@teamforrest.com>
Message-ID: <502E69E5.3030903@gmail.com>

Hello,

it should be straightforward starting with 3.3.0 to run Kamailio behind 
a port forwarding NAT (like amazon ec2), using the 'advertise' property 
of the listen parameter:

http://www.kamailio.org/wiki/cookbooks/3.3.x/core#listen

For example, default config will stay 'unchanged' apart of adding 
'advertise' attribute with public IP/port to listen. If the RTP has to 
be proxied, then I expect there is some extra work to do with the 
control of the rtpproxy, but if freeswitch stays in the loop of all 
calls and does media relaying (so no rtpproxy need), everything should 
just go.

Cheers,
Daniel


On 8/17/12 5:21 PM, Fred Posner wrote:
> Hi Spencer,
>
> Is Kamailio also natted? If so, you may have some issues... if not, it should work great. I run a server like this as well... very happy with it.
>
> I gave up on kamailio/freeswitch behind nat. Well, didn't give up, just don't have the time to make it work.
>
> With best regards,
>
> Fred
> http://qxork.com
>
> On Aug 17, 2012, at 11:04 AM, SamyGo wrote:
>
>> Hi,
>>
>> You kind of sound a little different here. Are you saying that the REGISTRATIONs will be handled by Freeswitch but store the registration Data in Kamailio "location" table !?
>>
>> Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the one on integrating the Asterisk Realtime with kamailio, In that articular configuration file Kamailio Forwards/relays the Registration attempts to the Media-Server (FreeSwitch in your case).
>>
>> That will get your NAT thing handled atleast. Next thing is , if I'm right about your requirement, saving of Authenticated users in the Kamailio locations table and I'm not very sure about how to do this.
>>
>>
>> BR
>> Sammy
>>
>>
>> On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> Hello,
>> I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal.  Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints?
>>
>> My thought is something like this:
>>
>> natted endpoint -> kamailio -> freeswitch
>> If the response is 200OK, save the registration.
>>
>> Thanks,
>> Spencer
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From spencer at 5ninesolutions.com  Fri Aug 17 18:29:18 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 17 Aug 2012 09:29:18 -0700
Subject: [SR-Users] Kamailio NAT traversal
In-Reply-To: <CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>
References: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>
	<CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>
Message-ID: <D0A85C84-6414-4E50-8526-6A48EF8AA981@5ninesolutions.com>

Hi Sammy and Fred,

Basically I'm building a hosted PBX platform using a muti domain FreeSWITCH setup.  Freeswitch and Kamailio are on a public IP.  Previously all endpoints registered to Freeswitch directly which works great.  For scalability purposes, my thought was to use Kamailio in front of Freeswitch to handle the grunt work of far end NAT traversal and using rtpproxy for media proxy.  Here's where things get a bit complicated.  I don't have a way to handle auth directly from Kamailio in this particular setup as the application's db schema will not map to what I need to do multi-domain auth in Kamailio.  So I'm blindly forwarding everything not originating from Freeswitch to Freeswitch and dealing with auth there.  

Since the client endpoints are behind NAT, this gets interesting :-).  I need to ping all NATed endpoints to keep the connection open.  I can't just forward the REGISTER and let Freeswitch ping the devices because it can't get to them as they opened a pinhole to Kamailio.  I also need presence to work, and all presence is handled by Freeswitch as well.  So my thought is this:

Let Kamailo ping the endpoints.  This obviously requires a registration.  Since Freeswitch is the only piece with credentials, save every REGISTER and remove the ones that fail in a failure route.  I check for a not empty $au before saving and remove the ones that fail.  I'm using usrloc in memory only mode so I'm not sure about the extra load this would create.

I need to store the REGISTERs in both Kamailio and Freeswitch.  I rewrite the Contact before forwarding to Freeswitch so that it sends INVITEs back to Kamailio which can get to the endpoints.  I can't use Path as I need the domain in the RURI.

I forward SUBSCRIBEs to Freeswitch with the Contact rewritten to point to the registered user in Kamailio.  Freeswitch then sends NOTIFYs to Kamailio which can get to the endpoints.

I'm completely open to input on how I might improve this setup or flaws in my logic but it at least works :-)

Thanks,
Spencer



On Aug 17, 2012, at 8:04 AM, SamyGo wrote:

> Hi,
> 
> You kind of sound a little different here. Are you saying that the REGISTRATIONs will be handled by Freeswitch but store the registration Data in Kamailio "location" table !? 
> 
> Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the one on integrating the Asterisk Realtime with kamailio, In that articular configuration file Kamailio Forwards/relays the Registration attempts to the Media-Server (FreeSwitch in your case). 
> 
> That will get your NAT thing handled atleast. Next thing is , if I'm right about your requirement, saving of Authenticated users in the Kamailio locations table and I'm not very sure about how to do this.
> 
> 
> BR
> Sammy
> 
> 
> On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello,
> I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal.  Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints?
> 
> My thought is something like this:
> 
> natted endpoint -> kamailio -> freeswitch
> If the response is 200OK, save the registration.
> 
> Thanks,
> Spencer
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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From fred at teamforrest.com  Fri Aug 17 18:34:57 2012
From: fred at teamforrest.com (Fred Posner)
Date: Fri, 17 Aug 2012 12:34:57 -0400
Subject: [SR-Users] Kamailio NAT traversal
In-Reply-To: <D0A85C84-6414-4E50-8526-6A48EF8AA981@5ninesolutions.com>
References: <019D6FF9-387A-4006-9282-849F9A28CDCB@5ninesolutions.com>
	<CAJUJwthVvfc5Vs-PMF-o1igFQ+twkNwpMau-mF5w06T_+FegNg@mail.gmail.com>
	<D0A85C84-6414-4E50-8526-6A48EF8AA981@5ninesolutions.com>
Message-ID: <F39B9EC2-4150-4133-BB63-B7DDCF25B677@teamforrest.com>

On my set-up, I forward incoming dids to freeswitch for handling, and then forward requests from registered users to another.

There's a great tutorial from Daniel/asipto that should be a very good guide:

http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc



With best regards,

Fred
http://qxork.com







On Aug 17, 2012, at 12:29 PM, Spencer Thomason wrote:

> Hi Sammy and Fred,
> 
> Basically I'm building a hosted PBX platform using a muti domain FreeSWITCH setup.  Freeswitch and Kamailio are on a public IP.  Previously all endpoints registered to Freeswitch directly which works great.  For scalability purposes, my thought was to use Kamailio in front of Freeswitch to handle the grunt work of far end NAT traversal and using rtpproxy for media proxy.  Here's where things get a bit complicated.  I don't have a way to handle auth directly from Kamailio in this particular setup as the application's db schema will not map to what I need to do multi-domain auth in Kamailio.  So I'm blindly forwarding everything not originating from Freeswitch to Freeswitch and dealing with auth there.  
> 
> Since the client endpoints are behind NAT, this gets interesting :-).  I need to ping all NATed endpoints to keep the connection open.  I can't just forward the REGISTER and let Freeswitch ping the devices because it can't get to them as they opened a pinhole to Kamailio.  I also need presence to work, and all presence is handled by Freeswitch as well.  So my thought is this:
> 
> Let Kamailo ping the endpoints.  This obviously requires a registration.  Since Freeswitch is the only piece with credentials, save every REGISTER and remove the ones that fail in a failure route.  I check for a not empty $au before saving and remove the ones that fail.  I'm using usrloc in memory only mode so I'm not sure about the extra load this would create.
> 
> I need to store the REGISTERs in both Kamailio and Freeswitch.  I rewrite the Contact before forwarding to Freeswitch so that it sends INVITEs back to Kamailio which can get to the endpoints.  I can't use Path as I need the domain in the RURI.
> 
> I forward SUBSCRIBEs to Freeswitch with the Contact rewritten to point to the registered user in Kamailio.  Freeswitch then sends NOTIFYs to Kamailio which can get to the endpoints.
> 
> I'm completely open to input on how I might improve this setup or flaws in my logic but it at least works :-)
> 
> Thanks,
> Spencer
> 
> 
> 
> On Aug 17, 2012, at 8:04 AM, SamyGo wrote:
> 
>> Hi,
>> 
>> You kind of sound a little different here. Are you saying that the REGISTRATIONs will be handled by Freeswitch but store the registration Data in Kamailio "location" table !? 
>> 
>> Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the one on integrating the Asterisk Realtime with kamailio, In that articular configuration file Kamailio Forwards/relays the Registration attempts to the Media-Server (FreeSwitch in your case). 
>> 
>> That will get your NAT thing handled atleast. Next thing is , if I'm right about your requirement, saving of Authenticated users in the Kamailio locations table and I'm not very sure about how to do this.
>> 
>> 
>> BR
>> Sammy
>> 
>> 
>> On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> Hello,
>> I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal.  Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints?
>> 
>> My thought is something like this:
>> 
>> natted endpoint -> kamailio -> freeswitch
>> If the response is 200OK, save the registration.
>> 
>> Thanks,
>> Spencer



From lfurrea at simplecs.net  Fri Aug 17 18:48:54 2012
From: lfurrea at simplecs.net (Luis Fernando Urrea)
Date: Fri, 17 Aug 2012 10:48:54 -0600
Subject: [SR-Users] Kamailio packet sourcing
Message-ID: <CAAPaQ=iO2SjphBt6ux2cBUZxcU3mvn2hEHgex2mzx3hzfWNWjg@mail.gmail.com>

Hi all,

I am sort of a new comer to kamailio and I have a config in which Kamailio
listens on different IPs that correspond to SSH VPN tunnels. The config
handles a MULTIDOMAIN scenario with FreeSWITCH as multi tenant media server.

The situation I am facing is that when a REGISTERED user calls in through
one of the SSH tunnels to the corresponding listening IP, Kamailio does not
seem to source the INVITE that goes to FreeSWITCH from that IP since on FS
I use "destination_field=network_addr" to try to apply different dialplan
rules per domain and I see the call on FS matching a non corresponding dial
plan entry sourced with an IP of a different  SSH tunnel.

Do I have any other option to send the call to FS using some other field to
match to my convenience?

I hope the description is clear enough, otherwise I could elaborate if
required.

Your input is highly appreciated.

Regards,

Luis
-- 
O: 4000-1020
C: 6040-3624
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From abalashov at evaristesys.com  Fri Aug 17 18:51:23 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 17 Aug 2012 12:51:23 -0400
Subject: [SR-Users] Kamailio packet sourcing
In-Reply-To: <CAAPaQ=iO2SjphBt6ux2cBUZxcU3mvn2hEHgex2mzx3hzfWNWjg@mail.gmail.com>
References: <CAAPaQ=iO2SjphBt6ux2cBUZxcU3mvn2hEHgex2mzx3hzfWNWjg@mail.gmail.com>
Message-ID: <502E768B.4090103@evaristesys.com>

Have a look at the force_send_socket / $fs approach.

On 08/17/2012 12:48 PM, Luis Fernando Urrea wrote:

> Hi all,
>
> I am sort of a new comer to kamailio and I have a config in which
> Kamailio listens on different IPs that correspond to SSH VPN tunnels.
> The config handles a MULTIDOMAIN scenario with FreeSWITCH as multi
> tenant media server.
>
> The situation I am facing is that when a REGISTERED user calls in
> through one of the SSH tunnels to the corresponding listening IP,
> Kamailio does not seem to source the INVITE that goes to FreeSWITCH from
> that IP since on FS I use "destination_field=network_addr" to try to
> apply different dialplan rules per domain and I see the call on FS
> matching a non corresponding dial plan entry sourced with an IP of a
> different  SSH tunnel.
>
> Do I have any other option to send the call to FS using some other field
> to match to my convenience?
>
> I hope the description is clear enough, otherwise I could elaborate if
> required.
>
> Your input is highly appreciated.
>
> Regards,
>
> Luis
> --
> O: 4000-1020
> C: 6040-3624
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From carlos.ruizdiaz at gmail.com  Fri Aug 17 22:32:17 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Fri, 17 Aug 2012 16:32:17 -0400
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <CA+3gBckuke2dV9kQSEjb2F_MMvD=+0tJ2O3DTMuGkydsWcZ8jg@mail.gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
	<CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
	<502D584A.8070104@gmail.com>
	<CA+3gBcn7V4od7=ATHZUC7zgNwFvQSuZKy2vMyNjGE5fq5LotWg@mail.gmail.com>
	<502E0F3A.2040401@gmail.com>
	<CA+3gBckuke2dV9kQSEjb2F_MMvD=+0tJ2O3DTMuGkydsWcZ8jg@mail.gmail.com>
Message-ID: <CA+3gBc=8VVV6225NooggghrwnS_p6iYENC7EA8uZSLBQzbm=8Q@mail.gmail.com>

I can't set any of the attributes values using the assignment syntax.
Everytime I do:

$(branch(q)[1]) = "0.1";

I get

ERROR: <core> [lvalue.c:354]: setting pvar failed.

This is true for all attributes, nut just for "q".

Am I doing something wrong? I'm using Kamailio 3.2.3.

Regards.

Carlos.


On Fri, Aug 17, 2012 at 11:38 AM, Carlos Ruiz D?az <
carlos.ruizdiaz at gmail.com> wrote:

> I'll try that :).
>
> Thanks for your help.
>
> Carlos.
>
>
> On Fri, Aug 17, 2012 at 5:30 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>
>> On 8/16/12 10:50 PM, Carlos Ruiz D?az wrote:
>>
>>
>>
>> On Thu, Aug 16, 2012 at 4:30 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>
>>> On 8/16/12 9:15 PM, Carlos Ruiz D?az wrote:
>>>
>>> Looks like km_append_branch() doesn't have all of the signatures that
>>> the original append_branch() has. I can't add q values for serial/parallel
>>> forking with km_append_branch().
>>>
>>>  What signature are you looking for?
>>>
>>
>>  The latter is the one I'm looking for, but with the possibility of
>> using pseudovars.
>>
>>  append_branch() overloads:
>>
>>  - append_branch();
>> - append_branch("sip:a at abc.com");
>> - append_branch("sip:a at abc.com", "0.1");
>>
>>
>> I am considering adding the second parameter to km_append_branch() for
>> backward compatibility reasons.
>>
>> Alternative is to use $branch(...) variable to set the q after you called
>> append branch:
>>
>>
>> http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#branch_attributes
>>
>> Q has to be set via integer value, with range from 0 to 100 (0.1 would be
>> 10).
>>
>>
>>
>>
>>  Also, apparently there's no way of setting the RURI by calling a
>> function that does that.
>>
>>
>>   What do you mean here? I don't understand.
>>>
>>
>>  I'm was talking about setting the request-uri from a PV.
>>
>>  For example: seturi("$var(my_uri)");
>>
>>  But it works using assignation.
>>
>>  $ru = $var(my_uri);
>>
>>
>> The core functions were added before the existence of cfg variables, they
>> take only static strings as parameters. The way to do it is via assignments.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>>
>>
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>  I guess I'll have to do it using assignation on $ru pseudovariable and
>>> writing a route to manage the serial forking I need.
>>>
>>>  Thanks.
>>>
>>>  Carlos.
>>>
>>> On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az <
>>> carlos.ruizdiaz at gmail.com> wrote:
>>>
>>>> Great! I'll give it a try.
>>>>
>>>>  Thanks!
>>>>
>>>>  Carlos.
>>>>
>>>>
>>>> On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla <
>>>> miconda at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>>
>>>>>
>>>>> On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>>>>>
>>>>>> Hello guys,
>>>>>>
>>>>>> Why I can only append_branch() a SIP URI as a literal constant
>>>>>> string? Why not from a pseudovariable? It's the same for most of the core
>>>>>> functions.
>>>>>>
>>>>>> My intention is to do something like this:
>>>>>>
>>>>>> append_branch("$cnx3a(route1)");
>>>>>> t_relay();
>>>>>>
>>>>>> where cnx3a is a variable containing the SIP URIs.
>>>>>>
>>>>>> The only "fix" I have found for this is by doing it as follows:
>>>>>>
>>>>>> $ru = $cnx3a(route1);
>>>>>> append_branch();
>>>>>> t_relay();
>>>>>>
>>>>>> Is there any other (and more elegant) way?
>>>>>>
>>>>>
>>>>>  you can use km_append_branch:
>>>>>
>>>>> http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> --
>>>>> Daniel-Constantin Mierla - http://www.asipto.com
>>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>>>>> http://asipto.com/u/katu
>>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>>>>> http://asipto.com/u/kpw
>>>>>
>>>>>
>>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>
>>
>
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From carlos.ruizdiaz at gmail.com  Fri Aug 17 22:55:11 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Fri, 17 Aug 2012 16:55:11 -0400
Subject: [SR-Users] append_branch with pseudo variables
In-Reply-To: <CA+3gBc=8VVV6225NooggghrwnS_p6iYENC7EA8uZSLBQzbm=8Q@mail.gmail.com>
References: <CA+3gBck8B1_JUgo5JbUZYWstORYEG+TnbF+Qg6Z6q7HhrxBrTg@mail.gmail.com>
	<502D1C0D.7050901@gmail.com>
	<CA+3gBcmgUMr6gj=WOzhUMHw84CytPFn_zXiOPH9f4CZJVtswkQ@mail.gmail.com>
	<CA+3gBcnZq37AzE38qkES0LYc3z6yJ7_w4pES8G=KTVOuZcr=Fw@mail.gmail.com>
	<502D584A.8070104@gmail.com>
	<CA+3gBcn7V4od7=ATHZUC7zgNwFvQSuZKy2vMyNjGE5fq5LotWg@mail.gmail.com>
	<502E0F3A.2040401@gmail.com>
	<CA+3gBckuke2dV9kQSEjb2F_MMvD=+0tJ2O3DTMuGkydsWcZ8jg@mail.gmail.com>
	<CA+3gBc=8VVV6225NooggghrwnS_p6iYENC7EA8uZSLBQzbm=8Q@mail.gmail.com>
Message-ID: <CA+3gBck3NpNyTE1SQ7L0YCSkQ-70thE0GoU-zceHYnnqn4X=qg@mail.gmail.com>

I figured out what was happening.

By doing "$(branch(q)[1]) = "0.1";" I was trying to write into an non
existent branch since 1 is actually the second branch because the main
branch ($ru)  is not counted.

$(branch(q)[0]) = "0.1" // is the 1st.
$(branch(q)[1]) = "0.1" // is the 2nd.

Regards.

Carlos.

On Fri, Aug 17, 2012 at 4:32 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com
> wrote:

> I can't set any of the attributes values using the assignment syntax.
> Everytime I do:
>
> $(branch(q)[1]) = "0.1";
>
> I get
>
> ERROR: <core> [lvalue.c:354]: setting pvar failed.
>
> This is true for all attributes, nut just for "q".
>
> Am I doing something wrong? I'm using Kamailio 3.2.3.
>
> Regards.
>
> Carlos.
>
>
> On Fri, Aug 17, 2012 at 11:38 AM, Carlos Ruiz D?az <
> carlos.ruizdiaz at gmail.com> wrote:
>
>> I'll try that :).
>>
>> Thanks for your help.
>>
>> Carlos.
>>
>>
>> On Fri, Aug 17, 2012 at 5:30 AM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>
>>> On 8/16/12 10:50 PM, Carlos Ruiz D?az wrote:
>>>
>>>
>>>
>>> On Thu, Aug 16, 2012 at 4:30 PM, Daniel-Constantin Mierla <
>>> miconda at gmail.com> wrote:
>>>
>>>>
>>>> On 8/16/12 9:15 PM, Carlos Ruiz D?az wrote:
>>>>
>>>> Looks like km_append_branch() doesn't have all of the signatures that
>>>> the original append_branch() has. I can't add q values for serial/parallel
>>>> forking with km_append_branch().
>>>>
>>>>  What signature are you looking for?
>>>>
>>>
>>>  The latter is the one I'm looking for, but with the possibility of
>>> using pseudovars.
>>>
>>>  append_branch() overloads:
>>>
>>>  - append_branch();
>>> - append_branch("sip:a at abc.com");
>>> - append_branch("sip:a at abc.com", "0.1");
>>>
>>>
>>> I am considering adding the second parameter to km_append_branch() for
>>> backward compatibility reasons.
>>>
>>> Alternative is to use $branch(...) variable to set the q after you
>>> called append branch:
>>>
>>>
>>> http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#branch_attributes
>>>
>>> Q has to be set via integer value, with range from 0 to 100 (0.1 would
>>> be 10).
>>>
>>>
>>>
>>>
>>>  Also, apparently there's no way of setting the RURI by calling a
>>> function that does that.
>>>
>>>
>>>   What do you mean here? I don't understand.
>>>>
>>>
>>>  I'm was talking about setting the request-uri from a PV.
>>>
>>>  For example: seturi("$var(my_uri)");
>>>
>>>  But it works using assignation.
>>>
>>>  $ru = $var(my_uri);
>>>
>>>
>>> The core functions were added before the existence of cfg variables,
>>> they take only static strings as parameters. The way to do it is via
>>> assignments.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>>
>>>
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>>  I guess I'll have to do it using assignation on $ru pseudovariable and
>>>> writing a route to manage the serial forking I need.
>>>>
>>>>  Thanks.
>>>>
>>>>  Carlos.
>>>>
>>>> On Thu, Aug 16, 2012 at 2:17 PM, Carlos Ruiz D?az <
>>>> carlos.ruizdiaz at gmail.com> wrote:
>>>>
>>>>> Great! I'll give it a try.
>>>>>
>>>>>  Thanks!
>>>>>
>>>>>  Carlos.
>>>>>
>>>>>
>>>>> On Thu, Aug 16, 2012 at 12:13 PM, Daniel-Constantin Mierla <
>>>>> miconda at gmail.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 8/16/12 5:25 PM, Carlos Ruiz D?az wrote:
>>>>>>
>>>>>>> Hello guys,
>>>>>>>
>>>>>>> Why I can only append_branch() a SIP URI as a literal constant
>>>>>>> string? Why not from a pseudovariable? It's the same for most of the core
>>>>>>> functions.
>>>>>>>
>>>>>>> My intention is to do something like this:
>>>>>>>
>>>>>>> append_branch("$cnx3a(route1)");
>>>>>>> t_relay();
>>>>>>>
>>>>>>> where cnx3a is a variable containing the SIP URIs.
>>>>>>>
>>>>>>> The only "fix" I have found for this is by doing it as follows:
>>>>>>>
>>>>>>> $ru = $cnx3a(route1);
>>>>>>> append_branch();
>>>>>>> t_relay();
>>>>>>>
>>>>>>> Is there any other (and more elegant) way?
>>>>>>>
>>>>>>
>>>>>>  you can use km_append_branch:
>>>>>>
>>>>>> http://kamailio.org/docs/modules/stable/modules_k/kex.html#id2551404
>>>>>>
>>>>>> Cheers,
>>>>>> Daniel
>>>>>>
>>>>>> --
>>>>>> Daniel-Constantin Mierla - http://www.asipto.com
>>>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>>>>>> http://asipto.com/u/katu
>>>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>>>>>> http://asipto.com/u/kpw
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> --
>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>>
>>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
>>>
>>>
>>
>
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From ushacked at gmail.com  Sun Aug 19 13:25:10 2012
From: ushacked at gmail.com (Uri Shacked)
Date: Sun, 19 Aug 2012 14:25:10 +0300
Subject: [SR-Users] avp usage limit
Message-ID: <CAMMbDhRpZOoAoM5aazYR1bFONjmr0E4TVju2Gvh-oTP19_JSuQ@mail.gmail.com>

Hi,

I wonder, is there a limit for the amount of AVP's i can use?
Let's say i store 6-9 characters (string) each time i set an AVP.

Thanks,
Uri
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From ushacked at gmail.com  Sun Aug 19 13:35:39 2012
From: ushacked at gmail.com (Uri Shacked)
Date: Sun, 19 Aug 2012 14:35:39 +0300
Subject: [SR-Users] does AVP's stay availeable after the transaction was
	answered?
Message-ID: <CAMMbDhSr0msxSAg6ztbBgzr+XLcfx3qJWADmtjb+k-mat5jj2A@mail.gmail.com>

Hi,

I am trying to use an AVP the I set on the INVITE when I recieve the BYE.
for some strange reason, the AVP is NULL when I test it.
Any ideas?

Uri
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From jh at tutpro.com  Sun Aug 19 13:41:40 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Sun, 19 Aug 2012 14:41:40 +0300
Subject: [SR-Users] does AVP's stay availeable after the transaction
	was	answered?
In-Reply-To: <CAMMbDhSr0msxSAg6ztbBgzr+XLcfx3qJWADmtjb+k-mat5jj2A@mail.gmail.com>
References: <CAMMbDhSr0msxSAg6ztbBgzr+XLcfx3qJWADmtjb+k-mat5jj2A@mail.gmail.com>
Message-ID: <20528.53492.463062.726135@siika.tutpro.com>

Uri Shacked writes:

> I am trying to use an AVP the I set on the INVITE when I recieve the BYE.
> for some strange reason, the AVP is NULL when I test it.

it is not strange reason, since avps are transaction specific and bye
does not belong to the same transaction as invite.

-- juha


From carlos.ruizdiaz at gmail.com  Sun Aug 19 19:33:49 2012
From: carlos.ruizdiaz at gmail.com (=?ISO-8859-1?Q?Carlos_Ruiz_D=EDaz?=)
Date: Sun, 19 Aug 2012 13:33:49 -0400
Subject: [SR-Users] avp usage limit
In-Reply-To: <CAMMbDhRpZOoAoM5aazYR1bFONjmr0E4TVju2Gvh-oTP19_JSuQ@mail.gmail.com>
References: <CAMMbDhRpZOoAoM5aazYR1bFONjmr0E4TVju2Gvh-oTP19_JSuQ@mail.gmail.com>
Message-ID: <CA+3gBckq5F8x9XD8t_Am+5Ax1QUfQWRhaSu6nUMMySBmm=NHWg@mail.gmail.com>

I believe it is directly related to the amount of physical memory available
for the process. Anyways, you can always delete the AVP after using it
although this is automatically done after the transaction finishes.

Maybe increasing the maximum memory limit could help:
http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:private-memory-size

Regards.

Carlos.

On Sun, Aug 19, 2012 at 7:25 AM, Uri Shacked <ushacked at gmail.com> wrote:

> Hi,
>
> I wonder, is there a limit for the amount of AVP's i can use?
> Let's say i store 6-9 characters (string) each time i set an AVP.
>
> Thanks,
> Uri
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From vitalik.voip at gmail.com  Mon Aug 20 00:29:30 2012
From: vitalik.voip at gmail.com (Vitaliy Aleksandrov)
Date: Mon, 20 Aug 2012 01:29:30 +0300
Subject: [SR-Users] Request for advice,
	running SIP-Router in OpenVZ Container
In-Reply-To: <001201cd7c75$cde02890$69a079b0$@gmail.com>
References: <001201cd7c75$cde02890$69a079b0$@gmail.com>
Message-ID: <503168CA.7040309@gmail.com>

On 08/17/2012 03:42 PM, JR Richardson wrote:
>
> Hi All,
>
> I'm considering running Kamailio as a virtual machine, with such low 
> utilization, it doesn't seem to make sense to keep running it on a 
> physical host server.
>
> I've been virtualizing Asterisk PBX's for years and run a host of 
> other virtualized servers with OpenVZ, VMware and MS V-Server without 
> issues.
>
> I'd like any opinions or thoughts about pitfalls or caveats I should 
> be aware of specific to virtualized SIP-Router/Kamailio.
>
> Thanks in advanced.
>
> JR
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

It works great in VMware and XEN VMs without any issues.
In my experience i had only two problems with virtual environment:
- I/O (network, disc).
- timing precision.

Disc I/O can be a problem if you are going to write lots of logs and the 
VM running kamailio are connected to overloaded disc array or external 
storage. Use only async syslog mode otherwise kamailio process will be 
blocked in log operation and will be not available to process incoming 
sip traffic. So use fast drives and try to avoid time consuming 
operations (syslog, db queries, radius, etc...) during message 
processing as they block current process and kamailio has fixed number 
of running processes.

 From my point of view timing precision is not a problem for a sip-proxy 
as it's not a big deal to send some re-transmit few milliseconds 
earlier/later.

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From ushacked at gmail.com  Mon Aug 20 06:23:54 2012
From: ushacked at gmail.com (Uri Shacked)
Date: Mon, 20 Aug 2012 07:23:54 +0300
Subject: [SR-Users] does AVP's stay availeable after the transaction was
	answered?
Message-ID: <CAMMbDhS5XEVDhjw8cMKZL56_-Hk8-PZF3VTnsNML_OHRvRy9Bg@mail.gmail.com>

OK,

I understan the AVP's are per transaction.
How can I save info from the invite and use it on the bye?  $sht(...)?
$dlg(flag), $dlg(set) ?

Thanks,
Uri

>  juha writes
>it is not strange reason, since avps are transaction specific and bye
>does not belong to the same transaction as invite.

> juha

>>Uri Shacked writes:

>* >I am trying to use an AVP the I set on the INVITE when I recieve the
BYE.*
*> *>* for some strange reason, the AVP is NULL when I test it. *
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From abalashov at evaristesys.com  Mon Aug 20 06:29:47 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 20 Aug 2012 00:29:47 -0400
Subject: [SR-Users] does AVP's stay availeable after the transaction was
 answered?
In-Reply-To: <CAMMbDhS5XEVDhjw8cMKZL56_-Hk8-PZF3VTnsNML_OHRvRy9Bg@mail.gmail.com>
References: <CAMMbDhS5XEVDhjw8cMKZL56_-Hk8-PZF3VTnsNML_OHRvRy9Bg@mail.gmail.com>
Message-ID: <5031BD3B.2070409@evaristesys.com>

What you're looking for is a mechanism to create values that persist 
throughout the life of a dialog, not a transaction.  There are two ways 
to accomplish that:

1. Traditional way, somewhat kludgy: stuff the value into the 
Record-Route header, taking advantage of its persistence across 
sequential messages:

http://www.kamailio.org/docs/modules/3.3.x/modules_k/rr.html#add-rr-param-id

Then, fish it back out with check_route_param().

2. Dialog variables: a new concept, relatively recently introduced:

http://www.kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#id2556135

-- Alex

On 08/20/2012 12:23 AM, Uri Shacked wrote:

> OK,
> I understan the AVP's are per transaction.
> How can I save info from the invite and use it on the bye?  $sht(...)?
> $dlg(flag), $dlg(set) ?
> Thanks,
> Uri
>  >  juha writes
>  >it is not strange reason, since avps are transaction specific and bye
>  >does not belong to the same transaction as invite.
>
>  > juha
>  >>Uri Shacked writes:
>
>  >/>I am trying to use an AVP the I set on the INVITE when I recieve the
> BYE./
> /> />/for some strange reason, the AVP is NULL when I test it. /
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From ibk at labhijau.net  Mon Aug 20 06:37:49 2012
From: ibk at labhijau.net (Iwan Budi Kusnanto)
Date: Mon, 20 Aug 2012 11:37:49 +0700
Subject: [SR-Users] avp usage limit
In-Reply-To: <CAMMbDhRpZOoAoM5aazYR1bFONjmr0E4TVju2Gvh-oTP19_JSuQ@mail.gmail.com>
References: <CAMMbDhRpZOoAoM5aazYR1bFONjmr0E4TVju2Gvh-oTP19_JSuQ@mail.gmail.com>
Message-ID: <CALv1wjFpThefdPhj683fSfRa1KQOqNz-3yD-sYeEtj_m-JCf1w@mail.gmail.com>

On Aug 19, 2012 6:25 PM, "Uri Shacked" <ushacked at gmail.com> wrote:

> Hi,
>
> I wonder, is there a limit for the amount of AVP's i can use?
> Let's say i store 6-9 characters (string) each time i set an AVP.
>
> Thanks,
> Uri
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From ronb at netweave.com  Mon Aug 20 01:43:06 2012
From: ronb at netweave.com (Ron Byer Jr)
Date: Sun, 19 Aug 2012 19:43:06 -0400
Subject: [SR-Users] UAC module
In-Reply-To: <5031747A.50307@netweave.com>
References: <502989A5.8000908@netweave.com> <5029F76D.6090903@gmail.com>
	<5031747A.50307@netweave.com>
Message-ID: <50317A0A.4040804@netweave.com>

How to subscribe 
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>

On 08/19/2012 07:19 PM, Ron Hopson wrote:
> Daniel,
>
> Thank you for your response.  I was away the remainder of the past week.
>
> Calls to uac_reg_lookup() and uac_reg_request_to(), with prior 
> assignment or no assignment, yield "fixing failed (code=-1)".  For 
> example,
>
>      if(uac_reg_request_to("$rU", 0)){        #line 76
>
>  0(16293) DEBUG: <core> [route.c:929]: fixing uac_reg_request_to()
>  0(16293) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
> cfg:/usr/local/etc/kamailio/kamailio.cfg:76
>
> I am most interested in function uac_reg_request_to().  I expect to 
> receive a REGISTER command from a local user and forward it with 
> proper credentials to a remote server.  I believe uac_reg_request_to() 
> will read (a memory image of) the uacreg file for the remote user's 
> name and credentials, and manage registration at the remote server.  
> Is this correct?
>
> I would like to lookup the registration information by $tU. Both of 
> the following code snippets get "fixing failed".
> (1)     if(uac_reg_request_to("$tU", 0)){
> and
> (2)        $rU = $tU;
>          if(uac_reg_request_to("$rU", 0)){
>
> Thank you for any hints or examples of how to use uac_reg_request_to().
>
> BTW:  How do I become a member of sr-users at lists.sip-router.org?
>
> regards,
> Ron Hopson
>
> On 8/14/2012 12:59 AM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> the parameters are expected to be variables, but looking at the log 
>> messages it seems that config optimization replace them with the 
>> string values, at you have the assignment just above the function, 
>> the parameters will be those string values always at runtime.
>>
>> Can you remove the assign statements and try again to see if starts ok?
>>
>> Cheers,
>> Daniel
>>
>> On 8/14/12 1:11 AM, Ron Hopson wrote:
>>> Hello,
>>>
>>> I cannot create a configuration consistent with uac_reg_lookup. Does 
>>> someone have a configuration with a simple example of uac_reg_lookup 
>>> that works with db_text file uacreg?
>>>
>>> I have kamailio 3.3 running on linux.  Based on suggestions in
>>> http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html
>>> and
>>> http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html 
>>>
>>> I created this record in db_text file uacreg:
>>> 2:12345678:daniel:mysipserver.com:daniel123:sipprovider.com:sipprovider.com:daniel123:xxxxx:"sip:sipprovider.com" 
>>>
>>>
>>> I assume that function uac_reg_lookup retrieves registration 
>>> information from this record after it is loaded into a memory table.
>>> One version of the configuration file is attached.  Here are three 
>>> examples of failures to create a configuration.
>>>
>>>     $rU = 12345678;
>>>
>>>     if(uac_reg_lookup("$rU", "$ru")){            #line 73
>>>
>>>  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 1
>>>  0(13856) DEBUG: <core> [rvalue.c:2861]: RV is int: 12345678
>>>  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>>>  0(13856) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>>>  0(13856) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>>> cfg:/usr/local/etc/kamailio/kamailio.cfg:73
>>> ---------
>>>     $rU = "12345678";
>>>
>>>     if(uac_reg_lookup("$rU", "$ru")){            #line 73
>>>
>>>  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>>>  0(13973) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
>>>  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>>>  0(13973) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>>>  0(13973) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>>> cfg:/usr/local/etc/kamailio/kamailio.cfg:73
>>> --------
>>>     $rU = "12345678";
>>>     $ru = "sip:mysipserver.com";
>>>
>>>     if(uac_reg_lookup("$rU", "$ru")){            #line 74
>>>
>>>
>>>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>>>  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
>>>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>>>  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: 
>>> "sip:mysipserver.com"
>>>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>>>  0(13976) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>>>  0(13976) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>>> cfg:/usr/local/etc/kamailio/kamailio.cfg:74
>>>
>>> ---------------------
>>> regards,
>>> Ron Hopson
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>> -- 
>> Daniel-Constantin Mierla -http://www.asipto.com
>> http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw
>

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From ronh at netweave.com  Mon Aug 20 01:19:22 2012
From: ronh at netweave.com (Ron Hopson)
Date: Sun, 19 Aug 2012 17:19:22 -0600
Subject: [SR-Users] UAC module
In-Reply-To: <5029F76D.6090903@gmail.com>
References: <502989A5.8000908@netweave.com> <5029F76D.6090903@gmail.com>
Message-ID: <5031747A.50307@netweave.com>

Daniel,

Thank you for your response.  I was away the remainder of the past week.

Calls to uac_reg_lookup() and uac_reg_request_to(), with prior 
assignment or no assignment, yield "fixing failed (code=-1)".  For example,

      if(uac_reg_request_to("$rU", 0)){        #line 76

  0(16293) DEBUG: <core> [route.c:929]: fixing uac_reg_request_to()
  0(16293) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
cfg:/usr/local/etc/kamailio/kamailio.cfg:76

I am most interested in function uac_reg_request_to().  I expect to 
receive a REGISTER command from a local user and forward it with proper 
credentials to a remote server.  I believe uac_reg_request_to() will 
read (a memory image of) the uacreg file for the remote user's name and 
credentials, and manage registration at the remote server.  Is this correct?

I would like to lookup the registration information by $tU. Both of the 
following code snippets get "fixing failed".
(1)     if(uac_reg_request_to("$tU", 0)){
and
(2)        $rU = $tU;
          if(uac_reg_request_to("$rU", 0)){

Thank you for any hints or examples of how to use uac_reg_request_to().

BTW:  How do I become a member of sr-users at lists.sip-router.org?

regards,
Ron Hopson

On 8/14/2012 12:59 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> the parameters are expected to be variables, but looking at the log 
> messages it seems that config optimization replace them with the 
> string values, at you have the assignment just above the function, the 
> parameters will be those string values always at runtime.
>
> Can you remove the assign statements and try again to see if starts ok?
>
> Cheers,
> Daniel
>
> On 8/14/12 1:11 AM, Ron Hopson wrote:
>> Hello,
>>
>> I cannot create a configuration consistent with uac_reg_lookup. Does 
>> someone have a configuration with a simple example of uac_reg_lookup 
>> that works with db_text file uacreg?
>>
>> I have kamailio 3.3 running on linux.  Based on suggestions in
>> http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html
>> and
>> http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html 
>>
>> I created this record in db_text file uacreg:
>> 2:12345678:daniel:mysipserver.com:daniel123:sipprovider.com:sipprovider.com:daniel123:xxxxx:"sip:sipprovider.com" 
>>
>>
>> I assume that function uac_reg_lookup retrieves registration 
>> information from this record after it is loaded into a memory table.
>> One version of the configuration file is attached.  Here are three 
>> examples of failures to create a configuration.
>>
>>     $rU = 12345678;
>>
>>     if(uac_reg_lookup("$rU", "$ru")){            #line 73
>>
>>  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 1
>>  0(13856) DEBUG: <core> [rvalue.c:2861]: RV is int: 12345678
>>  0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>>  0(13856) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>>  0(13856) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>> cfg:/usr/local/etc/kamailio/kamailio.cfg:73
>> ---------
>>     $rU = "12345678";
>>
>>     if(uac_reg_lookup("$rU", "$ru")){            #line 73
>>
>>  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>>  0(13973) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
>>  0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>>  0(13973) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>>  0(13973) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>> cfg:/usr/local/etc/kamailio/kamailio.cfg:73
>> --------
>>     $rU = "12345678";
>>     $ru = "sip:mysipserver.com";
>>
>>     if(uac_reg_lookup("$rU", "$ru")){            #line 74
>>
>>
>>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>>  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678"
>>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2
>>  0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: 
>> "sip:mysipserver.com"
>>  0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4
>>  0(13976) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup()
>>  0(13976) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>> cfg:/usr/local/etc/kamailio/kamailio.cfg:74
>>
>> ---------------------
>> regards,
>> Ron Hopson
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> -- 
> Daniel-Constantin Mierla -http://www.asipto.com
> http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw

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From miconda at gmail.com  Mon Aug 20 08:58:02 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 20 Aug 2012 08:58:02 +0200
Subject: [SR-Users] avp usage limit
In-Reply-To: <CA+3gBckq5F8x9XD8t_Am+5Ax1QUfQWRhaSu6nUMMySBmm=NHWg@mail.gmail.com>
References: <CAMMbDhRpZOoAoM5aazYR1bFONjmr0E4TVju2Gvh-oTP19_JSuQ@mail.gmail.com>
	<CA+3gBckq5F8x9XD8t_Am+5Ax1QUfQWRhaSu6nUMMySBmm=NHWg@mail.gmail.com>
Message-ID: <5031DFFA.3060600@gmail.com>

Hello,

indeed there is no limit set in the code for AVPs, it depends of the 
available memory, but it is shared memory. AVPs are not stored in 
private memory because they have to become available for replies, which 
can be handled by different processes.

Shared memory can be specified with -m command line parameter.

Note also that with the latest version, the size of private memory can 
be specified with -M command line parameter, there is no need to 
recompile anymore.

For more see 'kamailio -h'.

Cheers,
Daniel

On 8/19/12 7:33 PM, Carlos Ruiz D?az wrote:
> I believe it is directly related to the amount of physical memory 
> available for the process. Anyways, you can always delete the AVP 
> after using it although this is automatically done after the 
> transaction finishes.
>
> Maybe increasing the maximum memory limit could help: 
> http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:private-memory-size
>
> Regards.
>
> Carlos.
>
> On Sun, Aug 19, 2012 at 7:25 AM, Uri Shacked <ushacked at gmail.com 
> <mailto:ushacked at gmail.com>> wrote:
>
>     Hi,
>     I wonder, is there a limit for the amount of AVP's i can use?
>     Let's say i store 6-9 characters (string) each time i set an AVP.
>     Thanks,
>     Uri
>
>     _______________________________________________
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>     list
>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From miconda at gmail.com  Mon Aug 20 09:06:07 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 20 Aug 2012 09:06:07 +0200
Subject: [SR-Users] UAC module
In-Reply-To: <5031747A.50307@netweave.com>
References: <502989A5.8000908@netweave.com> <5029F76D.6090903@gmail.com>
	<5031747A.50307@netweave.com>
Message-ID: <5031E1DF.5060006@gmail.com>

Hello,


On 8/20/12 1:19 AM, Ron Hopson wrote:
> Daniel,
>
> Thank you for your response.  I was away the remainder of the past week.
>
> Calls to uac_reg_lookup() and uac_reg_request_to(), with prior 
> assignment or no assignment, yield "fixing failed (code=-1)".  For 
> example,
>
>      if(uac_reg_request_to("$rU", 0)){        #line 76
>
>  0(16293) DEBUG: <core> [route.c:929]: fixing uac_reg_request_to()
>  0(16293) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
> cfg:/usr/local/etc/kamailio/kamailio.cfg:76
>
> I am most interested in function uac_reg_request_to().  I expect to 
> receive a REGISTER command from a local user and forward it with 
> proper credentials to a remote server.  I believe uac_reg_request_to() 
> will read (a memory image of) the uacreg file for the remote user's 
> name and credentials, and manage registration at the remote server.  
> Is this correct?
>
> I would like to lookup the registration information by $tU. Both of 
> the following code snippets get "fixing failed".
> (1)     if(uac_reg_request_to("$tU", 0)){
> and
> (2)        $rU = $tU;
>          if(uac_reg_request_to("$rU", 0)){
>
> Thank you for any hints or examples of how to use uac_reg_request_to().
can you try putting the second parameter between double quotes? It 
should work without in the latest version, quotes being required in 
older releases, but just to check. I cannot test for the moment in my 
side, I will do it when I get a chance.

>
> BTW:  How do I become a member of sr-users at lists.sip-router.org?
>
See:

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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From klaus.mailinglists at pernau.at  Mon Aug 20 09:38:17 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Mon, 20 Aug 2012 09:38:17 +0200
Subject: [SR-Users] does AVP's stay availeable after the transaction was
 answered?
In-Reply-To: <CAMMbDhSr0msxSAg6ztbBgzr+XLcfx3qJWADmtjb+k-mat5jj2A@mail.gmail.com>
References: <CAMMbDhSr0msxSAg6ztbBgzr+XLcfx3qJWADmtjb+k-mat5jj2A@mail.gmail.com>
Message-ID: <5031E969.6040300@pernau.at>



On 19.08.2012 13:35, Uri Shacked wrote:
> Hi,
> I am trying to use an AVP the I set on the INVITE when I recieve the
> BYE. for some strange reason, the AVP is NULL when I test it.
> Any ideas?

The workarounds are:
- storing parameters in record-route cookies (parameter in Route header, 
public viewable)
- store variable in htable and reference it by a unique dialog-key, e.g. 
callid+fromtag

regards
Klaus


From miconda at gmail.com  Mon Aug 20 10:06:02 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 20 Aug 2012 10:06:02 +0200
Subject: [SR-Users] new feature: option to remove contacts based on sip nat
	keepalives
Message-ID: <5031EFEA.8080300@gmail.com>

Hello,

over the weekend I committed the code that allows to remove contacts 
from location table if the device does not respond to several attempts 
of SIP nat keepalives (usually OPTIONS requests).

The feature works only for SIP over UDP, being the only transport for 
which nathelper module sends keepalives (or sip ping requests as written 
in some docs).

Also, the lack of keepalive reply is tracked only for location records 
stored in memory (so it does not work with db only mode for usrloc). 
This is because the keepalives are sent stateless, not using tm module, 
to avoid overloading of timers and shared memory when dealing with large 
subscriber base -- therefore the implementation is very lightweight in 
using resources, targeting of course scalability. Detection of replies 
for keepalives is done based on from tag which carries the internal 
unique id of the location contact (which is available from v3.3).

In order to enable this feature you have to set the keepalive_timeout 
parameter of nathelper module -- that's the only addition you have to 
add to a config that sends SIP nat keepalives. Internally, the modules 
store the timestamp of the last SIP keepalive reply (nathelper detects 
the right reply and calls usrloc API for storing the timestamp in 
contact structure in memory).

http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244

Before sending a new keepalive to a contact, it is checking if the last 
keepalive reply is older than keepalive_timeout value and if yes, will 
lower the expires to 10 seconds (if the expires is higher). There is no 
special handling of keepalive reply codes, all of them will update the 
last keepalive timestamp.

This feature should be useful in mobile networks, to avoid stacking lot 
of invalid contacts which could result in lot of branches causing many 
retransmissions.

There is plant to go for tcp/tls as well and detect when the connection 
is down, but looks like no spare time in the near future. However, right 
now using set_forward_no_connect() will avoid trying to create new 
connections for natted uses whose registration connection is down.

Testing and feedback will be welcome.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw



From oej at edvina.net  Mon Aug 20 10:13:12 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Mon, 20 Aug 2012 10:13:12 +0200
Subject: [SR-Users] [sr-dev] new feature: option to remove contacts
	based on sip nat keepalives
In-Reply-To: <5031EFEA.8080300@gmail.com>
References: <5031EFEA.8080300@gmail.com>
Message-ID: <1CD23BC5-1002-4A6E-B42B-09C73530DB11@edvina.net>


20 aug 2012 kl. 10:06 skrev Daniel-Constantin Mierla:

> Hello,
> 
> over the weekend I committed the code that allows to remove contacts from location table if the device does not respond to several attempts of SIP nat keepalives (usually OPTIONS requests).
> 
> The feature works only for SIP over UDP, being the only transport for which nathelper module sends keepalives (or sip ping requests as written in some docs).
> 
> Also, the lack of keepalive reply is tracked only for location records stored in memory (so it does not work with db only mode for usrloc). This is because the keepalives are sent stateless, not using tm module, to avoid overloading of timers and shared memory when dealing with large subscriber base -- therefore the implementation is very lightweight in using resources, targeting of course scalability. Detection of replies for keepalives is done based on from tag which carries the internal unique id of the location contact (which is available from v3.3).
> 
> In order to enable this feature you have to set the keepalive_timeout parameter of nathelper module -- that's the only addition you have to add to a config that sends SIP nat keepalives. Internally, the modules store the timestamp of the last SIP keepalive reply (nathelper detects the right reply and calls usrloc API for storing the timestamp in contact structure in memory).
> 
> http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244
> 
> Before sending a new keepalive to a contact, it is checking if the last keepalive reply is older than keepalive_timeout value and if yes, will lower the expires to 10 seconds (if the expires is higher). There is no special handling of keepalive reply codes, all of them will update the last keepalive timestamp.
> 
> This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions.
> 
> There is plant to go for tcp/tls as well and detect when the connection is down, but looks like no spare time in the near future. However, right now using set_forward_no_connect() will avoid trying to create new connections for natted uses whose registration connection is down.
> 
On my personal wish list is to have an event route when a registered device becomes unreachable - like a closed TCP connection for a device behind NAT or failure to response to NAT pings.

/O

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From davemac2626 at live.co.uk  Mon Aug 20 13:51:57 2012
From: davemac2626 at live.co.uk (dave mccann)
Date: Mon, 20 Aug 2012 12:51:57 +0100
Subject: [SR-Users] Siremis login issue
Message-ID: <DUB117-W1040D3073CA62A3AE627CF787B90@phx.gbl>


Hi All

I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error:

"Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype, 
T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, 
T0.create_time, T0.update_by, T0.update_time FROM  `user` T0  WHERE 
T0.username='openser' and status='1'  LIMIT 1. SQLSTATE[42S02]: Base 
table or view not found: 1146 Table 'siremis.user' doesn't exist"

I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro').  I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared.  I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files.

Can anyone please help as this is that last bit that I need to overcome :)

Many thanks and best regards.

Dave
 		 	   		  
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From mm at 42com.com  Mon Aug 20 14:42:56 2012
From: mm at 42com.com (=?ISO-8859-1?Q?Max_M=FChlbronner?=)
Date: Mon, 20 Aug 2012 14:42:56 +0200
Subject: [SR-Users] Siremis login issue
In-Reply-To: <DUB117-W1040D3073CA62A3AE627CF787B90@phx.gbl>
References: <DUB117-W1040D3073CA62A3AE627CF787B90@phx.gbl>
Message-ID: <503230D0.9060007@42com.com>

On 08/20/2012 01:51 PM, dave mccann wrote:
> Base table or view not found: 1146 Table 'siremis.user' doesn't exist"

I did not use siremis before, but the error indicates a problem with the 
DB setup. (table missing)

Did you install/configure Database access correctly?


Best Regards


Max M.


From yufei.tao at redembedded.com  Mon Aug 20 15:35:02 2012
From: yufei.tao at redembedded.com (Yufei Tao)
Date: Mon, 20 Aug 2012 14:35:02 +0100
Subject: [SR-Users] [sr-dev] new feature: option to remove,
 contacts	based on sip nat keepalives
In-Reply-To: <mailman.5457.1345450405.2421.sr-users@lists.sip-router.org>
References: <mailman.5457.1345450405.2421.sr-users@lists.sip-router.org>
Message-ID: <50323D06.3020808@redembedded.com>

That's a very useful feature for updating presence information on behalf
of some clients that do not support presence, or simply crashed.
I will be watching this space for TCP/TLS support :)

Is there any other existing way to achieve this for clients over TCP/TLS?

Cheers,
Yufei

On 20/08/12 09:13, sr-users-request at lists.sip-router.org wrote:
> ------------------------------ Message: 5 Date: Mon, 20 Aug 2012
> 10:13:12 +0200 From: "Olle E. Johansson" <oej at edvina.net> Subject: Re:
> [SR-Users] [sr-dev] new feature: option to remove contacts based on
> sip nat keepalives To: miconda at gmail.com, Development mailing list of
> the sip-router project <sr-dev at lists.sip-router.org> Cc: "SIP Router -
> Kamailio \(OpenSER\) and SIP Express Router \(SER\) - Users Mailing
> List" <sr-users at lists.sip-router.org> Message-ID:
> <1CD23BC5-1002-4A6E-B42B-09C73530DB11 at edvina.net> Content-Type:
> text/plain; charset="us-ascii" 20 aug 2012 kl. 10:06 skrev
> Daniel-Constantin Mierla:
>> > Hello,
>> >
>> > over the weekend I committed the code that allows to remove contacts from location table if the device does not respond to several attempts of SIP nat keepalives (usually OPTIONS requests).
>> >
>> > The feature works only for SIP over UDP, being the only transport for which nathelper module sends keepalives (or sip ping requests as written in some docs).
>> >
>> > Also, the lack of keepalive reply is tracked only for location records stored in memory (so it does not work with db only mode for usrloc). This is because the keepalives are sent stateless, not using tm module, to avoid overloading of timers and shared memory when dealing with large subscriber base -- therefore the implementation is very lightweight in using resources, targeting of course scalability. Detection of replies for keepalives is done based on from tag which carries the internal unique id of the location contact (which is available from v3.3).
>> >
>> > In order to enable this feature you have to set the keepalive_timeout parameter of nathelper module -- that's the only addition you have to add to a config that sends SIP nat keepalives. Internally, the modules store the timestamp of the last SIP keepalive reply (nathelper detects the right reply and calls usrloc API for storing the timestamp in contact structure in memory).
>> >
>> > http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244
>> >
>> > Before sending a new keepalive to a contact, it is checking if the last keepalive reply is older than keepalive_timeout value and if yes, will lower the expires to 10 seconds (if the expires is higher). There is no special handling of keepalive reply codes, all of them will update the last keepalive timestamp.
>> >
>> > This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions.
>> >
>> > There is plant to go for tcp/tls as well and detect when the connection is down, but looks like no spare time in the near future. However, right now using set_forward_no_connect() will avoid trying to create new connections for natted uses whose registration connection is down.
>> >
> On my personal wish list is to have an event route when a registered device becomes unreachable - like a closed TCP connection for a device behind NAT or failure to response to NAT pings.
>
> /O

--
Yufei Tao
Red Embedded

This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message.

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From edesibe at googlemail.com  Mon Aug 20 15:00:52 2012
From: edesibe at googlemail.com (Ivan Milivojevic)
Date: Mon, 20 Aug 2012 15:00:52 +0200
Subject: [SR-Users] Siremis login issue
In-Reply-To: <DUB117-W1040D3073CA62A3AE627CF787B90@phx.gbl>
References: <DUB117-W1040D3073CA62A3AE627CF787B90@phx.gbl>
Message-ID: <00f701cd7ed3$d56b2320$80416960$@com>

H Dave, 
 
Do you have siremis.user table?
 
If you don't,  you should make it.
 
CREATE TABLE `users` (
  `SYSID` int(11) NOT NULL auto_increment,
  `USERID` varchar(15) NOT NULL,
  `PASSWORD` varchar(15) default NULL,
  PRIMARY KEY  (`SYSID`)
);
 
insert into ob_users(USERID, PASSWORD) values ('admin', 'admin');
 
And then try to access through web.
 
Regards,
Ivan
 
 
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of dave mccann
Sent: Monday, August 20, 2012 1:52 PM
To: sr-users at lists.sip-router.org
Subject: [SR-Users] Siremis login issue
 
Hi All

I have installed Kamailio with Siremis, Kamailio is working fine and starts
and I have been through the initial setup for Siremis however when i go to
login to the web management page it gives an error:

"Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype,
T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by,
T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE
T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base table or
view not found: 1146 Table 'siremis.user' doesn't exist"

I have tried the default "admin/admin" login as mentioned in the install
guide and have also tried the default passwords for Kamailio "openser -
(with default password 'openserrw') and openserro - (with default password
'openserro').  I have also tried root/*password* and also siremis/siremis
which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however
none were successful and the same error appeared.  I read that it was a
permissions error however have set directory /var/www/siremis-3.2.1 to chmod
777 including all sub dirs and files.

Can anyone please help as this is that last bit that I need to overcome :)

Many thanks and best regards.

Dave
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From ronh at netweave.com  Mon Aug 20 16:11:48 2012
From: ronh at netweave.com (Ron Hopson)
Date: Mon, 20 Aug 2012 08:11:48 -0600
Subject: [SR-Users] UAC module
In-Reply-To: <5031E1DF.5060006@gmail.com>
References: <502989A5.8000908@netweave.com> <5029F76D.6090903@gmail.com>
	<5031747A.50307@netweave.com> <5031E1DF.5060006@gmail.com>
Message-ID: <503245A4.1030704@netweave.com>

Daniel,

Adding double quotes around the second parameter of uac_reg_request_to 
does not help.

route{
     if (method=="REGISTER") {
         $rU = "daniel123";
         if(uac_reg_request_to("$rU", "0")){
             xlog("uac_reg_request_to remote user [$rU] on [$rd] via 
[$du]\n");
             t_on_failure("REMOTE_AUTH");
             t_relay();
         }
         exit;
     }
}
  0(24774) DEBUG: <core> [route.c:929]: fixing uac_reg_request_to()
  0(24774) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
cfg:/usr/local/etc/kamailio/kamailio.cfg:71

regards,
Ron


On 8/20/2012 1:06 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
>
> On 8/20/12 1:19 AM, Ron Hopson wrote:
>> Daniel,
>>
>> Thank you for your response.  I was away the remainder of the past week.
>>
>> Calls to uac_reg_lookup() and uac_reg_request_to(), with prior 
>> assignment or no assignment, yield "fixing failed (code=-1)". For 
>> example,
>>
>>      if(uac_reg_request_to("$rU", 0)){        #line 76
>>
>>  0(16293) DEBUG: <core> [route.c:929]: fixing uac_reg_request_to()
>>  0(16293) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at 
>> cfg:/usr/local/etc/kamailio/kamailio.cfg:76
>>
>> I am most interested in function uac_reg_request_to().  I expect to 
>> receive a REGISTER command from a local user and forward it with 
>> proper credentials to a remote server.  I believe 
>> uac_reg_request_to() will read (a memory image of) the uacreg file 
>> for the remote user's name and credentials, and manage registration 
>> at the remote server.  Is this correct?
>>
>> I would like to lookup the registration information by $tU. Both of 
>> the following code snippets get "fixing failed".
>> (1)     if(uac_reg_request_to("$tU", 0)){
>> and
>> (2)        $rU = $tU;
>>          if(uac_reg_request_to("$rU", 0)){
>>
>> Thank you for any hints or examples of how to use uac_reg_request_to().
> can you try putting the second parameter between double quotes? It 
> should work without in the latest version, quotes being required in 
> older releases, but just to check. I cannot test for the moment in my 
> side, I will do it when I get a chance.
>
>>
>> BTW:  How do I become a member of sr-users at lists.sip-router.org?
>>
> See:
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> Cheers,
> Daniel
>
> -- 
> Daniel-Constantin Mierla -http://www.asipto.com
> http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw

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From snagroot at yahoo.co.uk  Mon Aug 20 11:40:33 2012
From: snagroot at yahoo.co.uk (Marty Beyers)
Date: Mon, 20 Aug 2012 10:40:33 +0100 (BST)
Subject: [SR-Users] Send/Receive IMs
Message-ID: <1345455633.25589.YahooMailNeo@web29501.mail.ird.yahoo.com>

Hey guys. I am very new to Kamailio. Basically, at this time, I want to generate a SIP message (from an SMS) within Kannel, and send this message to my Kamailio server, which will forward the SIP message to another SIP server. Therefore my Kamailio Sip server would act as a single user agent connecting to only one other SIP server. So what I am unsure of is:
1. Is there a Kamailio interface (such as an HTTP server) which Kannel can connect to in order for Kannel to send multiple messages to my Kamailio Server, which in turn would be forwarded as SIP instant messages to another SIP server?
2. Which Kamailio api can I use to formulate and send the SIP instant messages??
3. Which Kamailio callback function could be called when I receive acknowledgement responses to the SIP IMs I have sent? 
4. Which Kamailio callback function could be called when I receive new SIP instant messages sent to me? 


It would be great if someone could help point me in the right direction. 

Many thanks.
Fred
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From davemac2626 at live.co.uk  Mon Aug 20 16:36:39 2012
From: davemac2626 at live.co.uk (dave mccann)
Date: Mon, 20 Aug 2012 15:36:39 +0100
Subject: [SR-Users] Siremis login issue
In-Reply-To: <mailman.5484.1345472242.2421.sr-users@lists.sip-router.org>
References: <mailman.5484.1345472242.2421.sr-users@lists.sip-router.org>
Message-ID: <DUB117-W140BAA9A0F2C9184A00BA5587B90@phx.gbl>


Hi All

I managed to figure out that the table had not been created in the database.  So i deleted the install.lock file in the Siremis directory to run the install again.  This time I clicked on all the boxes that were on step 2:  

 Create Siremis Database - this create database 'siremis'
 Import Default Data - this imports initial data into database 'siremis'
 Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules
 Replace DB Config - this updates Siremis configuration file with the details to access databases
Then after it had finished I was able to login using the default admin/admin.  Thanks for the reply Max greatly appreciated for the time taken to do so - Ivan - I didnt get chance to test what you had put however I will take a note of it for future reference.......hopefully I may be able to return the favor in the future if you guys need a hand.

Cheers guys
Dave


> From: sr-users-request at lists.sip-router.org
> Subject: sr-users Digest, Vol 87, Issue 64
> To: sr-users at lists.sip-router.org
> Date: Mon, 20 Aug 2012 16:17:22 +0200
> 
> Send sr-users mailing list submissions to
> 	sr-users at lists.sip-router.org
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> or, via email, send a message with subject or body 'help' to
> 	sr-users-request at lists.sip-router.org
> 
> You can reach the person managing the list at
> 	sr-users-owner at lists.sip-router.org
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of sr-users digest..."
> 
> 
> Today's Topics:
> 
>    1. Siremis login issue (dave mccann)
>    2. Re: Siremis login issue (Max M?hlbronner)
>    3. Re: [sr-dev] new feature: option to remove, contacts	based on
>       sip nat keepalives (Yufei Tao)
>    4. Re: Siremis login issue (Ivan Milivojevic)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Mon, 20 Aug 2012 12:51:57 +0100
> From: dave mccann <davemac2626 at live.co.uk>
> Subject: [SR-Users] Siremis login issue
> To: <sr-users at lists.sip-router.org>
> Message-ID: <DUB117-W1040D3073CA62A3AE627CF787B90 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> 
> Hi All
> 
> I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error:
> 
> "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype, 
> T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, 
> T0.create_time, T0.update_by, T0.update_time FROM  `user` T0  WHERE 
> T0.username='openser' and status='1'  LIMIT 1. SQLSTATE[42S02]: Base 
> table or view not found: 1146 Table 'siremis.user' doesn't exist"
> 
> I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro').  I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared.  I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files.
> 
> Can anyone please help as this is that last bit that I need to overcome :)
> 
> Many thanks and best regards.
> 
> Dave
>  		 	   		  
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> 
> ------------------------------
> 
> Message: 2
> Date: Mon, 20 Aug 2012 14:42:56 +0200
> From: Max M?hlbronner <mm at 42com.com>
> Subject: Re: [SR-Users] Siremis login issue
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> 	Users	Mailing List" <sr-users at lists.sip-router.org>
> Message-ID: <503230D0.9060007 at 42com.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> On 08/20/2012 01:51 PM, dave mccann wrote:
> > Base table or view not found: 1146 Table 'siremis.user' doesn't exist"
> 
> I did not use siremis before, but the error indicates a problem with the 
> DB setup. (table missing)
> 
> Did you install/configure Database access correctly?
> 
> 
> Best Regards
> 
> 
> Max M.
> 
> 
> 
> ------------------------------
> 
> 
> Message: 4
> Date: Mon, 20 Aug 2012 15:00:52 +0200
> From: Ivan Milivojevic <edesibe at googlemail.com>
> Subject: Re: [SR-Users] Siremis login issue
> To: "'SIP Router - Kamailio \(OpenSER\) and SIP Express Router \(SER\)
> 	-	Users	Mailing List'" <sr-users at lists.sip-router.org>
> Message-ID: <00f701cd7ed3$d56b2320$80416960$@com>
> Content-Type: text/plain; charset="iso-8859-2"
> 
> H Dave, 
>  
> Do you have siremis.user table?
>  
> If you don't,  you should make it.
>  
> CREATE TABLE `users` (
>   `SYSID` int(11) NOT NULL auto_increment,
>   `USERID` varchar(15) NOT NULL,
>   `PASSWORD` varchar(15) default NULL,
>   PRIMARY KEY  (`SYSID`)
> );
>  
> insert into ob_users(USERID, PASSWORD) values ('admin', 'admin');
>  
> And then try to access through web.
>  
> Regards,
> Ivan
>  
>  
> From: sr-users-bounces at lists.sip-router.org
> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of dave mccann
> Sent: Monday, August 20, 2012 1:52 PM
> To: sr-users at lists.sip-router.org
> Subject: [SR-Users] Siremis login issue
>  
> Hi All
> 
> I have installed Kamailio with Siremis, Kamailio is working fine and starts
> and I have been through the initial setup for Siremis however when i go to
> login to the web management page it gives an error:
> 
> "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype,
> T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by,
> T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE
> T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base table or
> view not found: 1146 Table 'siremis.user' doesn't exist"
> 
> I have tried the default "admin/admin" login as mentioned in the install
> guide and have also tried the default passwords for Kamailio "openser -
> (with default password 'openserrw') and openserro - (with default password
> 'openserro').  I have also tried root/*password* and also siremis/siremis
> which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however
> none were successful and the same error appeared.  I read that it was a
> permissions error however have set directory /var/www/siremis-3.2.1 to chmod
> 777 including all sub dirs and files.
> 
> Can anyone please help as this is that last bit that I need to overcome :)
> 
> Many thanks and best regards.
> 
> Dave
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> ------------------------------
> 
> _______________________________________________
> sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> 
> End of sr-users Digest, Vol 87, Issue 64
> ****************************************
 		 	   		  
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From spencer at 5ninesolutions.com  Mon Aug 20 23:26:35 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 20 Aug 2012 14:26:35 -0700
Subject: [SR-Users] SUBSCRIBE/NOTIFY record route
Message-ID: <22b26f342aad1aa1544f7019d5126199@mail.5ninesolutions.com>

  

Hello all, 

I'm using Kamailio (on a public IP) as a border proxy
that handles NAT traversal (detection and keep alives) for several
Freeswitch servers which have public IPs as well. I'm having a problem
record-routing SUBSCRIBEs and NOTIFYs for nated UACs. If the UAC is
behind NAT, I am adding a Record-Route to the initial SUBSCRIBE and
NOTIFY request. This works great until the UAC (in this case a Cisco
SPA509G) refreshes the subcription and sends the SUBSCRIBE directly to
the Freeswitch server. Since the request doesn't pass through the proxy,
Freeswitch then sends NOTIFYs directly to the UAC which only recieves
the first NOTIFY as the NAT pinhole is still open from the SUBSCRIBE.
The UAC did receive a NOTIFY validating the subscription, and continues
to refresh the subscription directly to the freeswitch server. The
completely breaks presence. 

What is the correct way to ensure all
SUBSCRIBE/NOTIFYs pass through the proxy? I have found that I can force
this behavior by modifying the Contact header in the 202 Accepted to
point back to the proxy but I'm not sure if this is what I "should" be
doing. 

Thanks for any assistance and please forgive my ignorance when
is comes to SIP presence! 

Spencer 
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From klaus.mailinglists at pernau.at  Tue Aug 21 09:47:19 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 21 Aug 2012 09:47:19 +0200
Subject: [SR-Users] SUBSCRIBE/NOTIFY record route
In-Reply-To: <22b26f342aad1aa1544f7019d5126199@mail.5ninesolutions.com>
References: <22b26f342aad1aa1544f7019d5126199@mail.5ninesolutions.com>
Message-ID: <50333D07.1000802@pernau.at>



On 20.08.2012 23:26, Spencer Thomason wrote:
> Hello all,
>
> I'm using Kamailio (on a public IP) as a border proxy that handles NAT
> traversal (detection and keep alives) for several Freeswitch servers
> which have public IPs as well.  I'm having a problem record-routing
> SUBSCRIBEs and NOTIFYs for nated UACs.  If the UAC is behind NAT, I am
> adding a Record-Route to the initial SUBSCRIBE and NOTIFY request.  This
> works great until the UAC (in this case a Cisco SPA509G) refreshes the
> subcription and sends the SUBSCRIBE directly to the Freeswitch server.

If the phone receives the Record-Route headers then this is a bug in the 
Cisco phone.

>   Since the request doesn't pass through the proxy, Freeswitch then
> sends NOTIFYs directly to the UAC which only recieves the first NOTIFY
> as the NAT pinhole is still open from the SUBSCRIBE.  The UAC did

I think this is a bug in Freeswitch. IIRC the route set can not be 
changed during a dialog.

> receive a NOTIFY validating the subscription, and continues to refresh
> the subscription directly to the freeswitch server.  The completely
> breaks presence.
>
> What is the correct way to ensure all SUBSCRIBE/NOTIFYs pass through the
> proxy?  I have found that I can force this behavior by modifying the
> Contact header in the 202 Accepted to point back to the proxy but I'm
> not sure if this is what I "should" be doing.

That's what I did once because of this Cisco bug. I do not see another 
possibility. If the phone ignores the route-set but sends to the Contact 
URI, then you have to change the Contact URI to point to Kamailio, and 
on incoming reSUBSCRIBEs change the RURI to the original Contact URI 
(Freeswitch).

To rewrite the contact you have 2 possibilities:

a) if you have a fixed mapping from user to freeswitch-server, then 
apply this mapping also for the reSUBSCRIBE to get the IP address of the 
freeswitch node

b) if there is no fixed mapping, encode the original Contact URI into 
the new one. When receiving the reSUBSCRIBE, decode the RURI to get the 
original Contact URI back. If freeswitch does not care about the RURI in 
the reSUBSCRIBE then this can be done quite easy (by using the nathelper 
functions), e.g.:


reply-route of SUBSCRIBE:

   ...
   remove_hf("Contact");
   # '1' means UDP transport
   append_hf("Contact: <sip:ip.addr.of.proxy:port;alias=$si~$sp~1>\r\n");
   ...

WITHIN_DIALOG route:
   ...
   if (has_totag()) {
     if (is_method("SUBSCRIBE")) {
       handle_ruri_alias();
     }
   }
   ...

regards
Klaus



From serusers at iptel.org  Tue Aug 21 10:02:08 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Tue, 21 Aug 2012 00:02:08 -0800
Subject: [SR-Users] vacancy #428
Message-ID: <48CE4DE81D28F0979F67D50F59851F61@nbhomcjhicinjndpyj.realliving.com>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  Collectively, we focus on different sources of power, new
  green tech, and reusable materials. We are advertising for someone from your
   region to be employed by us. We require: [...] 

Content analysis details:   (7.5 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
-0.0 SPF_PASS               SPF: sender matches SPF record
 0.0 T_FRT_PROFILE2         BODY: ReplaceTags: Profile (2)
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
                            [109.100.26.103 listed in zen.spamhaus.org]
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [109.100.26.103 listed in bb.barracudacentral.org]
 1.7 URIBL_DBL_SPAM         Contains an URL listed in the DBL blocklist
                            [URIs: euroinjobs.com]
 1.7 URIBL_BLACK            Contains an URL listed in the URIBL blacklist
                            [URIs: euroinjobs.com]
-1.9 BAYES_00               BODY: Bayes spam probability is 0 to 1%
                            [score: 0.0000]
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS


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From david.patino at quobis.com  Tue Aug 21 10:59:36 2012
From: david.patino at quobis.com (=?ISO-8859-1?Q?David_Pati=F1o?=)
Date: Tue, 21 Aug 2012 10:59:36 +0200
Subject: [SR-Users] Websocket INVITE 407 error
Message-ID: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>

Hi there,

I'm new on kamailio users list.
Now I'm developing a client in javascript (using websockets) to SIP calls.

I'm using last devel version of kamailio with websocket support. I've
already get a correct REGISTER transaction but now I'm not able to get
a correct INVITE transaction. Actually I'm not sure if this a kamailio
error or a problem with my source code however I'm using the same
function to get digest info from received nonce both REGISTER and
INVITE transactions. The INVITE transaction always enters in an
endless loop with 407 requests.

Please, is there anybody who could help me?

Those are the frames of my code (obtained from chrome browser console):

wsUri: ws://10.1.20.40:80/
SOCKET CREATED
SOCKET OPENED

SENT:
REGISTER sip:10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
To: sip:101 at 10.1.20.40
Call-ID: 0EQSBkoG9RGgfg3E
CSeq: 1 REGISTER
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
	;reg-id=98
	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"


RESPONSE:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
Call-ID: 0EQSBkoG9RGgfg3E
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
Server: kamailio (3.4.0-dev3 (i386/linux))
Content-Length: 0


SENT:
REGISTER sip:10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
To: sip:101 at 10.1.20.40
Call-ID: 0EQSBkoG9RGgfg3E
CSeq: 2 REGISTER
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
	;reg-id=98
	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
Authorization: Digest username="101", realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"


RESPONSE:
SIP/2.0 200 OK
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
Call-ID: 0EQSBkoG9RGgfg3E
CSeq: 2 REGISTER
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
Server: kamailio (3.4.0-dev3 (i386/linux))
Content-Length: 0


SENT:
INVITE sip:102 at 10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40
Call-ID: 9KoKnFdh285k2jg4
CSeq: 1 INVITE
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
Content-Type: application/sdp


RESPONSE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
Call-ID: 9KoKnFdh285k2jg4
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
Server: kamailio (3.4.0-dev3 (i386/linux))
Content-Length: 0


SENT:
ACK sip:102 at 10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
Call-ID: 9KoKnFdh285k2jg4
CSeq: 1 ACK
Max-Forwards: 70
Supported: path, outbound, gruu


SENT:
INVITE sip:102 at 10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40
Call-ID: 9KoKnFdh285k2jg4
CSeq: 2 INVITE
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
Proxy-Authorization: Digest username="101", realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
Content-Type: application/sdp


RESPONSE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
Call-ID: 9KoKnFdh285k2jg4
CSeq: 2 INVITE
Proxy-Authenticate: Digest realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
Server: kamailio (3.4.0-dev3 (i386/linux))
Content-Length: 0


SENT:
ACK sip:102 at 10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
Call-ID: 9KoKnFdh285k2jg4
CSeq: 2 ACK
Max-Forwards: 70
Supported: path, outbound, gruu


SENT:
INVITE sip:102 at 10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40
Call-ID: 9KoKnFdh285k2jg4
CSeq: 3 INVITE
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
Proxy-Authorization: Digest username="101", realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
Content-Type: application/sdp


RESPONSE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
Call-ID: 9KoKnFdh285k2jg4
CSeq: 3 INVITE
Proxy-Authenticate: Digest realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
Server: kamailio (3.4.0-dev3 (i386/linux))
Content-Length: 0


.......and so on in an endless loop with 407 requests

Thanks a lot

Kind Regards

--
     --DAVID--


From peter.dunkley at crocodile-rcs.com  Tue Aug 21 13:04:50 2012
From: peter.dunkley at crocodile-rcs.com (Peter Dunkley)
Date: Tue, 21 Aug 2012 12:04:50 +0100
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
Message-ID: <1345547090.13632.2.camel@pd-notebook-linux.croc.internal>

Hello,

This does look like an issue with authentication rather than WebSockets.
Have you tried using an ordinary SIP client (for example, Jitsi) with
your Kamailio configuration to verify that the authentication part of
kamailio.cfg is correct?

Regards,

Peter

On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:

> Hi there,
> 
> I'm new on kamailio users list.
> Now I'm developing a client in javascript (using websockets) to SIP calls.
> 
> I'm using last devel version of kamailio with websocket support. I've
> already get a correct REGISTER transaction but now I'm not able to get
> a correct INVITE transaction. Actually I'm not sure if this a kamailio
> error or a problem with my source code however I'm using the same
> function to get digest info from received nonce both REGISTER and
> INVITE transactions. The INVITE transaction always enters in an
> endless loop with 407 requests.
> 
> Please, is there anybody who could help me?
> 
> Those are the frames of my code (obtained from chrome browser console):
> 
> wsUri: ws://10.1.20.40:80/
> SOCKET CREATED
> SOCKET OPENED
> 
> SENT:
> REGISTER sip:10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
> To: sip:101 at 10.1.20.40
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 1 REGISTER
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
> 	;reg-id=98
> 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
> 
> 
> RESPONSE:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 1 REGISTER
> WWW-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
> 
> 
> SENT:
> REGISTER sip:10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
> To: sip:101 at 10.1.20.40
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
> 	;reg-id=98
> 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
> Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
> 
> 
> RESPONSE:
> SIP/2.0 200 OK
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 2 REGISTER
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
> 
> 
> SENT:
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 1 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Content-Type: application/sdp
> 
> 
> RESPONSE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 1 INVITE
> Proxy-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
> 
> 
> SENT:
> ACK sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 1 ACK
> Max-Forwards: 70
> Supported: path, outbound, gruu
> 
> 
> SENT:
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 2 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> Content-Type: application/sdp
> 
> 
> RESPONSE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 2 INVITE
> Proxy-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
> 
> 
> SENT:
> ACK sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 2 ACK
> Max-Forwards: 70
> Supported: path, outbound, gruu
> 
> 
> SENT:
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 3 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> Content-Type: application/sdp
> 
> 
> RESPONSE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 3 INVITE
> Proxy-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
> 
> 
> .......and so on in an endless loop with 407 requests
> 
> Thanks a lot
> 
> Kind Regards
> 
> --
>      --DAVID--
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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From david.patino at quobis.com  Tue Aug 21 13:24:27 2012
From: david.patino at quobis.com (=?ISO-8859-1?Q?David_Pati=F1o?=)
Date: Tue, 21 Aug 2012 13:24:27 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
Message-ID: <CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>

Hello,

I've just test a sip call from an X-Lite client to Ekiga client an
everything has gone ok.
It seems an issue with authentication over WebSockets but I have no
experience in this area so I can not say where the problem is (and
REGISTER auth works fine).

Any other idea or anything else to check?

I'm blocked since last week.

Thanks a lot.
Kind Regards,

--
     --DAVID--




2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com>
>
> Hello,
>
> This does look like an issue with authentication rather than WebSockets.
> Have you tried using an ordinary SIP client (for example, Jitsi) with your
> Kamailio configuration to verify that the authentication part of
> kamailio.cfg is correct?
>
> Regards,
>
> Peter
>
>
> On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
>
> Hi there,
>
> I'm new on kamailio users list.
> Now I'm developing a client in javascript (using websockets) to SIP calls.
>
> I'm using last devel version of kamailio with websocket support. I've
> already get a correct REGISTER transaction but now I'm not able to get
> a correct INVITE transaction. Actually I'm not sure if this a kamailio
> error or a problem with my source code however I'm using the same
> function to get digest info from received nonce both REGISTER and
> INVITE transactions. The INVITE transaction always enters in an
> endless loop with 407 requests.
>
> Please, is there anybody who could help me?
>
> Those are the frames of my code (obtained from chrome browser console):
>
> wsUri: ws://10.1.20.40:80/
> SOCKET CREATED
> SOCKET OPENED
>
> SENT:
> REGISTER sip:10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
> To: sip:101 at 10.1.20.40
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 1 REGISTER
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
> 	;reg-id=98
> 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>
>
> RESPONSE:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/WS
>
> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 1 REGISTER
> WWW-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
>
>
> SENT:
> REGISTER sip:10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
> To: sip:101 at 10.1.20.40
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
> 	;reg-id=98
> 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
> Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
>
>
> RESPONSE:
> SIP/2.0 200 OK
> Via: SIP/2.0/WS
>
> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
> Call-ID: 0EQSBkoG9RGgfg3E
> CSeq: 2 REGISTER
> Contact:
> <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
>
>
> SENT:
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 1 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Content-Type: application/sdp
>
>
> RESPONSE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/WS
>
> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 1 INVITE
> Proxy-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
>
>
> SENT:
> ACK sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 1 ACK
> Max-Forwards: 70
> Supported: path, outbound, gruu
>
>
> SENT:
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 2 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> Content-Type: application/sdp
>
>
> RESPONSE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/WS
>
> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 2 INVITE
> Proxy-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
>
>
> SENT:
> ACK sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 2 ACK
> Max-Forwards: 70
> Supported: path, outbound, gruu
>
>
> SENT:
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 3 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> Content-Type: application/sdp
>
>
> RESPONSE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/WS
>
> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 3 INVITE
> Proxy-Authenticate: Digest realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
> Re"
> Server: kamailio (3.4.0-dev3 (i386/linux))
> Content-Length: 0
>
>
> .......and so on in an endless loop with 407 requests
>
> Thanks a lot
>
> Kind Regards
>
> --
>      --DAVID--
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From peter.dunkley at crocodile-rcs.com  Tue Aug 21 13:29:49 2012
From: peter.dunkley at crocodile-rcs.com (Peter Dunkley)
Date: Tue, 21 Aug 2012 12:29:49 +0100
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
Message-ID: <1345548589.13632.4.camel@pd-notebook-linux.croc.internal>

Hi,

If the authentication works with another client then it suggest a
problem with your HTML5 client.

There is nothing special or different about authenticating over a
WebSocket connection.

Another thing would be to try a different WebSocket client, for example
sipml5 (which is known to work), to check.

Regards,

Peter

On Tue, 2012-08-21 at 13:24 +0200, David Pati?o wrote:

> Hello,
> 
> I've just test a sip call from an X-Lite client to Ekiga client an
> everything has gone ok.
> It seems an issue with authentication over WebSockets but I have no
> experience in this area so I can not say where the problem is (and
> REGISTER auth works fine).
> 
> Any other idea or anything else to check?
> 
> I'm blocked since last week.
> 
> Thanks a lot.
> Kind Regards,
> 
> --
>      --DAVID--
> 
> 
> 
> 
> 2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com>
> >
> > Hello,
> >
> > This does look like an issue with authentication rather than WebSockets.
> > Have you tried using an ordinary SIP client (for example, Jitsi) with your
> > Kamailio configuration to verify that the authentication part of
> > kamailio.cfg is correct?
> >
> > Regards,
> >
> > Peter
> >
> >
> > On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
> >
> > Hi there,
> >
> > I'm new on kamailio users list.
> > Now I'm developing a client in javascript (using websockets) to SIP calls.
> >
> > I'm using last devel version of kamailio with websocket support. I've
> > already get a correct REGISTER transaction but now I'm not able to get
> > a correct INVITE transaction. Actually I'm not sure if this a kamailio
> > error or a problem with my source code however I'm using the same
> > function to get digest info from received nonce both REGISTER and
> > INVITE transactions. The INVITE transaction always enters in an
> > endless loop with 407 requests.
> >
> > Please, is there anybody who could help me?
> >
> > Those are the frames of my code (obtained from chrome browser console):
> >
> > wsUri: ws://10.1.20.40:80/
> > SOCKET CREATED
> > SOCKET OPENED
> >
> > SENT:
> > REGISTER sip:10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
> > From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
> > To: sip:101 at 10.1.20.40
> > Call-ID: 0EQSBkoG9RGgfg3E
> > CSeq: 1 REGISTER
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
> > 	;reg-id=98
> > 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
> >
> >
> > RESPONSE:
> > SIP/2.0 401 Unauthorized
> > Via: SIP/2.0/WS
> >
> > KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
> > From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
> > To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
> > Call-ID: 0EQSBkoG9RGgfg3E
> > CSeq: 1 REGISTER
> > WWW-Authenticate: Digest realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> > Server: kamailio (3.4.0-dev3 (i386/linux))
> > Content-Length: 0
> >
> >
> > SENT:
> > REGISTER sip:10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
> > From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
> > To: sip:101 at 10.1.20.40
> > Call-ID: 0EQSBkoG9RGgfg3E
> > CSeq: 2 REGISTER
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
> > 	;reg-id=98
> > 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
> > Authorization: Digest username="101", realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> > uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
> >
> >
> > RESPONSE:
> > SIP/2.0 200 OK
> > Via: SIP/2.0/WS
> >
> > KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
> > From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
> > To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
> > Call-ID: 0EQSBkoG9RGgfg3E
> > CSeq: 2 REGISTER
> > Contact:
> > <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
> > Server: kamailio (3.4.0-dev3 (i386/linux))
> > Content-Length: 0
> >
> >
> > SENT:
> > INVITE sip:102 at 10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 1 INVITE
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> > 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> > Content-Type: application/sdp
> >
> >
> > RESPONSE:
> > SIP/2.0 407 Proxy Authentication Required
> > Via: SIP/2.0/WS
> >
> > KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 1 INVITE
> > Proxy-Authenticate: Digest realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> > Server: kamailio (3.4.0-dev3 (i386/linux))
> > Content-Length: 0
> >
> >
> > SENT:
> > ACK sip:102 at 10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 1 ACK
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> >
> >
> > SENT:
> > INVITE sip:102 at 10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 2 INVITE
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> > 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> > Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> > uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> > Content-Type: application/sdp
> >
> >
> > RESPONSE:
> > SIP/2.0 407 Proxy Authentication Required
> > Via: SIP/2.0/WS
> >
> > KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 2 INVITE
> > Proxy-Authenticate: Digest realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
> > Server: kamailio (3.4.0-dev3 (i386/linux))
> > Content-Length: 0
> >
> >
> > SENT:
> > ACK sip:102 at 10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 2 ACK
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> >
> >
> > SENT:
> > INVITE sip:102 at 10.1.20.40 SIP/2.0
> > Via: SIP/2.0/WS
> > KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 3 INVITE
> > Max-Forwards: 70
> > Supported: path, outbound, gruu
> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
> > 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> > Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> > uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> > Content-Type: application/sdp
> >
> >
> > RESPONSE:
> > SIP/2.0 407 Proxy Authentication Required
> > Via: SIP/2.0/WS
> >
> > KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
> > From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> > To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
> > Call-ID: 9KoKnFdh285k2jg4
> > CSeq: 3 INVITE
> > Proxy-Authenticate: Digest realm="10.1.20.40",
> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
> > Re"
> > Server: kamailio (3.4.0-dev3 (i386/linux))
> > Content-Length: 0
> >
> >
> > .......and so on in an endless loop with 407 requests
> >
> > Thanks a lot
> >
> > Kind Regards
> >
> > --
> >      --DAVID--
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> >
> > --
> > Peter Dunkley
> > Technical Director
> > Crocodile RCS Ltd
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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From miconda at gmail.com  Tue Aug 21 14:51:00 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Aug 2012 14:51:00 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <1345548589.13632.4.camel@pd-notebook-linux.croc.internal>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
	<1345548589.13632.4.camel@pd-notebook-linux.croc.internal>
Message-ID: <50338434.6090600@gmail.com>

Hello,

you should run kamailio in debug mode and watch the log messages to see 
what is not ok with the authentication. Maybe that gives you some hints.

Cheers,
Daniel


On 8/21/12 1:29 PM, Peter Dunkley wrote:
> Hi,
>
> If the authentication works with another client then it suggest a 
> problem with your HTML5 client.
>
> There is nothing special or different about authenticating over a 
> WebSocket connection.
>
> Another thing would be to try a different WebSocket client, for 
> example sipml5 (which is known to work), to check.
>
> Regards,
>
> Peter
>
> On Tue, 2012-08-21 at 13:24 +0200, David Pati?o wrote:
>> Hello,
>>
>> I've just test a sip call from an X-Lite client to Ekiga client an
>> everything has gone ok.
>> It seems an issue with authentication over WebSockets but I have no
>> experience in this area so I can not say where the problem is (and
>> REGISTER auth works fine).
>>
>> Any other idea or anything else to check?
>>
>> I'm blocked since last week.
>>
>> Thanks a lot.
>> Kind Regards,
>>
>> --
>>       --DAVID--
>>
>>
>>
>>
>> 2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com  <mailto:peter.dunkley at crocodile-rcs.com>>
>> >
>> > Hello,
>> >
>> > This does look like an issue with authentication rather than WebSockets.
>> > Have you tried using an ordinary SIP client (for example, Jitsi) with your
>> > Kamailio configuration to verify that the authentication part of
>> > kamailio.cfg is correct?
>> >
>> > Regards,
>> >
>> > Peter
>> >
>> >
>> > On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
>> >
>> > Hi there,
>> >
>> > I'm new on kamailio users list.
>> > Now I'm developing a client in javascript (using websockets) to SIP calls.
>> >
>> > I'm using last devel version of kamailio with websocket support. I've
>> > already get a correct REGISTER transaction but now I'm not able to get
>> > a correct INVITE transaction. Actually I'm not sure if this a kamailio
>> > error or a problem with my source code however I'm using the same
>> > function to get digest info from received nonce both REGISTER and
>> > INVITE transactions. The INVITE transaction always enters in an
>> > endless loop with 407 requests.
>> >
>> > Please, is there anybody who could help me?
>> >
>> > Those are the frames of my code (obtained from chrome browser console):
>> >
>> > wsUri: ws://10.1.20.40:80/
>> > SOCKET CREATED
>> > SOCKET OPENED
>> >
>> > SENT:
>> > REGISTERsip:10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
>> > From:sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>> > To:sip:101 at 10.1.20.40
>> > Call-ID: 0EQSBkoG9RGgfg3E
>> > CSeq: 1 REGISTER
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>> > 	;reg-id=98
>> > 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>> >
>> >
>> > RESPONSE:
>> > SIP/2.0 401 Unauthorized
>> > Via: SIP/2.0/WS
>> >
>> > KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
>> > From:sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>> > To:sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
>> > Call-ID: 0EQSBkoG9RGgfg3E
>> > CSeq: 1 REGISTER
>> > WWW-Authenticate: Digest realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>> > Server: kamailio (3.4.0-dev3 (i386/linux))
>> > Content-Length: 0
>> >
>> >
>> > SENT:
>> > REGISTERsip:10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
>> > From:sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>> > To:sip:101 at 10.1.20.40
>> > Call-ID: 0EQSBkoG9RGgfg3E
>> > CSeq: 2 REGISTER
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>> > 	;reg-id=98
>> > 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>> > Authorization: Digest username="101", realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>> > uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
>> >
>> >
>> > RESPONSE:
>> > SIP/2.0 200 OK
>> > Via: SIP/2.0/WS
>> >
>> > KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
>> > From:sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>> > To:sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
>> > Call-ID: 0EQSBkoG9RGgfg3E
>> > CSeq: 2 REGISTER
>> > Contact:
>> > <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
>> > Server: kamailio (3.4.0-dev3 (i386/linux))
>> > Content-Length: 0
>> >
>> >
>> > SENT:
>> > INVITEsip:102 at 10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 1 INVITE
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>> > 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>> > Content-Type: application/sdp
>> >
>> >
>> > RESPONSE:
>> > SIP/2.0 407 Proxy Authentication Required
>> > Via: SIP/2.0/WS
>> >
>> > KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 1 INVITE
>> > Proxy-Authenticate: Digest realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>> > Server: kamailio (3.4.0-dev3 (i386/linux))
>> > Content-Length: 0
>> >
>> >
>> > SENT:
>> > ACKsip:102 at 10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 1 ACK
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> >
>> >
>> > SENT:
>> > INVITEsip:102 at 10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 2 INVITE
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>> > 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>> > Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>> > uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>> > Content-Type: application/sdp
>> >
>> >
>> > RESPONSE:
>> > SIP/2.0 407 Proxy Authentication Required
>> > Via: SIP/2.0/WS
>> >
>> > KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 2 INVITE
>> > Proxy-Authenticate: Digest realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>> > Server: kamailio (3.4.0-dev3 (i386/linux))
>> > Content-Length: 0
>> >
>> >
>> > SENT:
>> > ACKsip:102 at 10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 2 ACK
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> >
>> >
>> > SENT:
>> > INVITEsip:102 at 10.1.20.40  SIP/2.0
>> > Via: SIP/2.0/WS
>> > KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 3 INVITE
>> > Max-Forwards: 70
>> > Supported: path, outbound, gruu
>> > Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>> > 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>> > Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>> > uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>> > Content-Type: application/sdp
>> >
>> >
>> > RESPONSE:
>> > SIP/2.0 407 Proxy Authentication Required
>> > Via: SIP/2.0/WS
>> >
>> > KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
>> > From:sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> > To:sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
>> > Call-ID: 9KoKnFdh285k2jg4
>> > CSeq: 3 INVITE
>> > Proxy-Authenticate: Digest realm="10.1.20.40",
>> > nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
>> > Re"
>> > Server: kamailio (3.4.0-dev3 (i386/linux))
>> > Content-Length: 0
>> >
>> >
>> > .......and so on in an endless loop with 407 requests
>> >
>> > Thanks a lot
>> >
>> > Kind Regards
>> >
>> > --
>> >      --DAVID--
>> >
>> > _______________________________________________
>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> >sr-users at lists.sip-router.org  <mailto:sr-users at lists.sip-router.org>
>> >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >
>> >
>> > --
>> > Peter Dunkley
>> > Technical Director
>> > Crocodile RCS Ltd
>> >
>> >
>> > _______________________________________________
>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> >sr-users at lists.sip-router.org  <mailto:sr-users at lists.sip-router.org>
>> >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> >
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org  <mailto:sr-users at lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> -- 
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 05-08, 2012 - http://asipto.com/u/kat

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From klaus.mailinglists at pernau.at  Tue Aug 21 14:59:56 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 21 Aug 2012 14:59:56 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
Message-ID: <5033864C.1070904@pernau.at>

I just hacked a small PHP script to calculate and verify the nonce 
manually:
http://pernau.at/kd/sipdigest.php

regards
Klaus

PS: do not use "secret" passwords for testing :-)



On 21.08.2012 13:24, David Pati?o wrote:
> Hello,
>
> I've just test a sip call from an X-Lite client to Ekiga client an
> everything has gone ok.
> It seems an issue with authentication over WebSockets but I have no
> experience in this area so I can not say where the problem is (and
> REGISTER auth works fine).
>
> Any other idea or anything else to check?
>
> I'm blocked since last week.
>
> Thanks a lot.
> Kind Regards,
>
> --
>       --DAVID--
>
>
>
>
> 2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com>
>>
>> Hello,
>>
>> This does look like an issue with authentication rather than WebSockets.
>> Have you tried using an ordinary SIP client (for example, Jitsi) with your
>> Kamailio configuration to verify that the authentication part of
>> kamailio.cfg is correct?
>>
>> Regards,
>>
>> Peter
>>
>>
>> On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
>>
>> Hi there,
>>
>> I'm new on kamailio users list.
>> Now I'm developing a client in javascript (using websockets) to SIP calls.
>>
>> I'm using last devel version of kamailio with websocket support. I've
>> already get a correct REGISTER transaction but now I'm not able to get
>> a correct INVITE transaction. Actually I'm not sure if this a kamailio
>> error or a problem with my source code however I'm using the same
>> function to get digest info from received nonce both REGISTER and
>> INVITE transactions. The INVITE transaction always enters in an
>> endless loop with 407 requests.
>>
>> Please, is there anybody who could help me?
>>
>> Those are the frames of my code (obtained from chrome browser console):
>>
>> wsUri: ws://10.1.20.40:80/
>> SOCKET CREATED
>> SOCKET OPENED
>>
>> SENT:
>> REGISTER sip:10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>> To: sip:101 at 10.1.20.40
>> Call-ID: 0EQSBkoG9RGgfg3E
>> CSeq: 1 REGISTER
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>> 	;reg-id=98
>> 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>
>>
>> RESPONSE:
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/WS
>>
>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
>> Call-ID: 0EQSBkoG9RGgfg3E
>> CSeq: 1 REGISTER
>> WWW-Authenticate: Digest realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>> Server: kamailio (3.4.0-dev3 (i386/linux))
>> Content-Length: 0
>>
>>
>> SENT:
>> REGISTER sip:10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>> To: sip:101 at 10.1.20.40
>> Call-ID: 0EQSBkoG9RGgfg3E
>> CSeq: 2 REGISTER
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>> 	;reg-id=98
>> 	;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>> Authorization: Digest username="101", realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>> uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
>>
>>
>> RESPONSE:
>> SIP/2.0 200 OK
>> Via: SIP/2.0/WS
>>
>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
>> Call-ID: 0EQSBkoG9RGgfg3E
>> CSeq: 2 REGISTER
>> Contact:
>> <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
>> Server: kamailio (3.4.0-dev3 (i386/linux))
>> Content-Length: 0
>>
>>
>> SENT:
>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 1 INVITE
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>> Content-Type: application/sdp
>>
>>
>> RESPONSE:
>> SIP/2.0 407 Proxy Authentication Required
>> Via: SIP/2.0/WS
>>
>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 1 INVITE
>> Proxy-Authenticate: Digest realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>> Server: kamailio (3.4.0-dev3 (i386/linux))
>> Content-Length: 0
>>
>>
>> SENT:
>> ACK sip:102 at 10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 1 ACK
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>>
>>
>> SENT:
>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 2 INVITE
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>> Content-Type: application/sdp
>>
>>
>> RESPONSE:
>> SIP/2.0 407 Proxy Authentication Required
>> Via: SIP/2.0/WS
>>
>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 2 INVITE
>> Proxy-Authenticate: Digest realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>> Server: kamailio (3.4.0-dev3 (i386/linux))
>> Content-Length: 0
>>
>>
>> SENT:
>> ACK sip:102 at 10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 2 ACK
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>>
>>
>> SENT:
>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>> Via: SIP/2.0/WS
>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 3 INVITE
>> Max-Forwards: 70
>> Supported: path, outbound, gruu
>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>> 	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>> Content-Type: application/sdp
>>
>>
>> RESPONSE:
>> SIP/2.0 407 Proxy Authentication Required
>> Via: SIP/2.0/WS
>>
>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
>> Call-ID: 9KoKnFdh285k2jg4
>> CSeq: 3 INVITE
>> Proxy-Authenticate: Digest realm="10.1.20.40",
>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
>> Re"
>> Server: kamailio (3.4.0-dev3 (i386/linux))
>> Content-Length: 0
>>
>>
>> .......and so on in an endless loop with 407 requests
>>
>> Thanks a lot
>>
>> Kind Regards
>>
>> --
>>       --DAVID--
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Peter Dunkley
>> Technical Director
>> Crocodile RCS Ltd
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From david.patino at quobis.com  Tue Aug 21 16:05:01 2012
From: david.patino at quobis.com (=?ISO-8859-1?Q?David_Pati=F1o?=)
Date: Tue, 21 Aug 2012 16:05:01 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <5033864C.1070904@pernau.at>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
	<5033864C.1070904@pernau.at>
Message-ID: <CABmM4BCcyDHhd69B289JaMS7nC6Zs+_0CmNdri-veLZojKiC9g@mail.gmail.com>

Hi,

Awesome tool!

I've been checking my info with the output of your script and the
output is the same. Maybe the auth problem is not here or maybe I've
made mistakes. Could you please tell my if the data I'm using is
correct?

For example, I'm user 101 and I want to call to user 102. So I've used
the following values to create the INVITE response to 407 message:

User: 101
Pass: 101
realm: 10.1.20.40
nonce:"UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv"
uri: sip:10.1.20.40

HA1 params: 101:10.1.20.40:101
HA1: 1d578520f175df632f0850415e603029
HA2 params: INVITE:sip:10.1.20.40
HA2: 793dd6fa2e181e25a226cc09efc6dc2c
Sol params: 1d578520f175df632f0850415e603029:UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv:793dd6fa2e181e25a226cc09efc6dc2c
Sol: 362206a07a3782c6e2557699abe5bf7b

Are those values correct?

Thank you very much.
Kind Regards,

--
     --DAVID--



2012/8/21 Klaus Darilion <klaus.mailinglists at pernau.at>:
> I just hacked a small PHP script to calculate and verify the nonce manually:
> http://pernau.at/kd/sipdigest.php
>
> regards
> Klaus
>
> PS: do not use "secret" passwords for testing :-)
>
>
>
>
> On 21.08.2012 13:24, David Pati?o wrote:
>>
>> Hello,
>>
>> I've just test a sip call from an X-Lite client to Ekiga client an
>> everything has gone ok.
>> It seems an issue with authentication over WebSockets but I have no
>> experience in this area so I can not say where the problem is (and
>> REGISTER auth works fine).
>>
>> Any other idea or anything else to check?
>>
>> I'm blocked since last week.
>>
>> Thanks a lot.
>> Kind Regards,
>>
>> --
>>       --DAVID--
>>
>>
>>
>>
>> 2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com>
>>>
>>>
>>> Hello,
>>>
>>> This does look like an issue with authentication rather than WebSockets.
>>> Have you tried using an ordinary SIP client (for example, Jitsi) with
>>> your
>>> Kamailio configuration to verify that the authentication part of
>>> kamailio.cfg is correct?
>>>
>>> Regards,
>>>
>>> Peter
>>>
>>>
>>> On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
>>>
>>> Hi there,
>>>
>>> I'm new on kamailio users list.
>>> Now I'm developing a client in javascript (using websockets) to SIP
>>> calls.
>>>
>>> I'm using last devel version of kamailio with websocket support. I've
>>> already get a correct REGISTER transaction but now I'm not able to get
>>> a correct INVITE transaction. Actually I'm not sure if this a kamailio
>>> error or a problem with my source code however I'm using the same
>>> function to get digest info from received nonce both REGISTER and
>>> INVITE transactions. The INVITE transaction always enters in an
>>> endless loop with 407 requests.
>>>
>>> Please, is there anybody who could help me?
>>>
>>> Those are the frames of my code (obtained from chrome browser console):
>>>
>>> wsUri: ws://10.1.20.40:80/
>>> SOCKET CREATED
>>> SOCKET OPENED
>>>
>>> SENT:
>>> REGISTER sip:10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
>>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>>> To: sip:101 at 10.1.20.40
>>> Call-ID: 0EQSBkoG9RGgfg3E
>>> CSeq: 1 REGISTER
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>>>         ;reg-id=98
>>>         ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>>
>>>
>>> RESPONSE:
>>> SIP/2.0 401 Unauthorized
>>> Via: SIP/2.0/WS
>>>
>>>
>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
>>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
>>> Call-ID: 0EQSBkoG9RGgfg3E
>>> CSeq: 1 REGISTER
>>> WWW-Authenticate: Digest realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>> Content-Length: 0
>>>
>>>
>>> SENT:
>>> REGISTER sip:10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
>>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>>> To: sip:101 at 10.1.20.40
>>> Call-ID: 0EQSBkoG9RGgfg3E
>>> CSeq: 2 REGISTER
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>>>         ;reg-id=98
>>>         ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>> Authorization: Digest username="101", realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>> uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
>>>
>>>
>>> RESPONSE:
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/WS
>>>
>>>
>>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
>>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
>>> Call-ID: 0EQSBkoG9RGgfg3E
>>> CSeq: 2 REGISTER
>>> Contact:
>>>
>>> <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>> Content-Length: 0
>>>
>>>
>>> SENT:
>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 1 INVITE
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>         ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>> Content-Type: application/sdp
>>>
>>>
>>> RESPONSE:
>>> SIP/2.0 407 Proxy Authentication Required
>>> Via: SIP/2.0/WS
>>>
>>>
>>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 1 INVITE
>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>> Content-Length: 0
>>>
>>>
>>> SENT:
>>> ACK sip:102 at 10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 1 ACK
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>>
>>>
>>> SENT:
>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 2 INVITE
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>         ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>>> Content-Type: application/sdp
>>>
>>>
>>> RESPONSE:
>>> SIP/2.0 407 Proxy Authentication Required
>>> Via: SIP/2.0/WS
>>>
>>>
>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 2 INVITE
>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>> Content-Length: 0
>>>
>>>
>>> SENT:
>>> ACK sip:102 at 10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 2 ACK
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>>
>>>
>>> SENT:
>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>> Via: SIP/2.0/WS
>>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 3 INVITE
>>> Max-Forwards: 70
>>> Supported: path, outbound, gruu
>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>         ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>>> Content-Type: application/sdp
>>>
>>>
>>> RESPONSE:
>>> SIP/2.0 407 Proxy Authentication Required
>>> Via: SIP/2.0/WS
>>>
>>>
>>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
>>> Call-ID: 9KoKnFdh285k2jg4
>>> CSeq: 3 INVITE
>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
>>> Re"
>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>> Content-Length: 0
>>>
>>>
>>> .......and so on in an endless loop with 407 requests
>>>
>>> Thanks a lot
>>>
>>> Kind Regards
>>>
>>> --
>>>       --DAVID--
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Peter Dunkley
>>> Technical Director
>>> Crocodile RCS Ltd
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>


From cezary.siwek at asergis.com  Tue Aug 21 16:14:30 2012
From: cezary.siwek at asergis.com (Cezary Siwek)
Date: Tue, 21 Aug 2012 15:14:30 +0100
Subject: [SR-Users] Send/Receive IMs
In-Reply-To: <1345455633.25589.YahooMailNeo@web29501.mail.ird.yahoo.com>
References: <1345455633.25589.YahooMailNeo@web29501.mail.ird.yahoo.com>
Message-ID: <503397C6.1080205@asergis.com>

Hi Marty/Fred,


On 20/08/2012 10:40, Marty Beyers wrote:
> Hey guys. I am very new to Kamailio. Basically, at this time, I want 
> to generate a SIP message (from an SMS) within Kannel, and send this 
> message to my Kamailio server, which will forward the SIP message to 
> another SIP server. Therefore my Kamailio Sip server would act as a 
> single user agent connecting to only one other SIP server. So what I 
> am unsure of is:
> 1. Is there a Kamailio interface (such as an HTTP server) which Kannel 
> can connect to in order for Kannel to send multiple messages to my 
> Kamailio Server, which in turn would be forwarded as SIP instant 
> messages to another SIP server?
You can probably use xhttp module but you will need to extract the GET 
variables from the URL. You can also an external program to send a 
MESSAGE via SIP (ie sipsak) or write your own interface script that sits 
between Kannel and  Kamailio;

> 2. Which Kamailio api can I use to formulate and send the SIP instant 
> messages?
check |uac_req_send() from the UAC module.
|
> 3. Which Kamailio callback function could be called when I receive 
> acknowledgement responses to the SIP IMs I have sent?

check |http_query()||| from the utils module;

> 4. Which Kamailio callback function could be called when I receive new 
> SIP instant messages sent to me?
>
http_query() again but you can also write your own function in Perl or 
LUA that interacts with the Kannel.

> It would be great if someone could help point me in the right direction.
> Many thanks.
> Fred
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



From bligthart at btlnet.co.uk  Tue Aug 21 17:45:01 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Tue, 21 Aug 2012 15:45:01 +0000
Subject: [SR-Users] db_cassandra
In-Reply-To: <502A5829.3090201@1and1.ro>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A46FE.7080403@1and1.ro>
	<D8261FE064C78B44B77EF7C4569D6CBF0D11AF60@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A5829.3090201@1and1.ro>
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11CD44@DB3PRD0510MB380.eurprd05.prod.outlook.com>

Hi,
  Kamailio now, connects to Cassandra ok. However the query against the column family
does not yield any results. Followed the docs to only create the column family inside the keyspace.


From serusers at iptel.org  Tue Aug 21 19:30:22 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Tue, 21 Aug 2012 18:30:22 +0100
Subject: [SR-Users] New Job position
Message-ID: <6969332266.6DH0E3CE998672@aarwooeskejt.ulmcteqa.ua>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  Collectively, we focus on different sources of power, new
  green tech, and reusable materials. We are advertising for someone from your
   region to be employed by us. We require: [...] 

Content analysis details:   (10.6 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
                            [91.54.224.170 listed in zen.spamhaus.org]
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
-0.0 SPF_PASS               SPF: sender matches SPF record
 0.0 T_FRT_PROFILE2         BODY: ReplaceTags: Profile (2)
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [91.54.224.170 listed in bb.barracudacentral.org]
 1.3 RCVD_IN_RP_RNBL        RBL: Relay in RNBL,
                            https://senderscore.org/blacklistlookup/
                            [91.54.224.170 listed in bl.score.senderscore.com]
 1.7 URIBL_DBL_SPAM         Contains an URL listed in the DBL blocklist
                            [URIs: euroinjobs.com]
 1.7 URIBL_BLACK            Contains an URL listed in the URIBL blacklist
                            [URIs: euroinjobs.com]
-1.9 BAYES_00               BODY: Bayes spam probability is 0 to 1%
                            [score: 0.0000]
 1.6 URIBL_SBL              Contains an URL listed in the SBL blocklist
                            [URIs: euroinjobs.com]
 1.0 RDNS_DYNAMIC           Delivered to internal network by host with
                            dynamic-looking rDNS


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From klaus.mailinglists at pernau.at  Tue Aug 21 19:42:45 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 21 Aug 2012 19:42:45 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <CABmM4BCcyDHhd69B289JaMS7nC6Zs+_0CmNdri-veLZojKiC9g@mail.gmail.com>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
	<5033864C.1070904@pernau.at>
	<CABmM4BCcyDHhd69B289JaMS7nC6Zs+_0CmNdri-veLZojKiC9g@mail.gmail.com>
Message-ID: <5033C895.9020605@pernau.at>

When you send the INVITE, then the RURI is not sip:10.1.20.40 but 
sip:102 at 10.1.20.40 - it includes the user part.

I see that you have it correct in the Proxy-Authorization header of the 
INVITE, but maybe you calculated the response with the wrong URI.

Anyway, taking a look at your INVITE from your previous email:

INVITE sip:102 at 10.1.20.40 SIP/2.0
Via: SIP/2.0/WS
KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102 at 10.1.20.40
Call-ID: 9KoKnFdh285k2jg4
CSeq: 3 INVITE
Max-Forwards: 70
Supported: path, outbound, gruu
Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
	;gr=urn:uuid:beb-68a9-6396e5f4;ob>
Proxy-Authorization: Digest username="101", realm="10.1.20.40",
nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
Content-Type: application/sdp

This gives as input to my script:
username: 101
password: 101 (hope this is correct)
realm: 10.1.20.40
nonce: UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re
uri: sip:102 at 10.1.20.40
method: INVITE

--> my script calculates 4a6dab565fa5f1e39e13960c2f323eab but the INVITE 
contains f4bbe128abf3acb3a4e11882749d8798. So, your stack seems to be buggy.

regards
klaus




On 21.08.2012 16:05, David Pati?o wrote:
> Hi,
>
> Awesome tool!
>
> I've been checking my info with the output of your script and the
> output is the same. Maybe the auth problem is not here or maybe I've
> made mistakes. Could you please tell my if the data I'm using is
> correct?
>
> For example, I'm user 101 and I want to call to user 102. So I've used
> the following values to create the INVITE response to 407 message:
>
> User: 101
> Pass: 101
> realm: 10.1.20.40
> nonce:"UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv"
> uri: sip:10.1.20.40
>
> HA1 params: 101:10.1.20.40:101
> HA1: 1d578520f175df632f0850415e603029
> HA2 params: INVITE:sip:10.1.20.40
> HA2: 793dd6fa2e181e25a226cc09efc6dc2c
> Sol params: 1d578520f175df632f0850415e603029:UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv:793dd6fa2e181e25a226cc09efc6dc2c
> Sol: 362206a07a3782c6e2557699abe5bf7b
>
> Are those values correct?
>
> Thank you very much.
> Kind Regards,
>
> --
>       --DAVID--
>
>
>
> 2012/8/21 Klaus Darilion <klaus.mailinglists at pernau.at>:
>> I just hacked a small PHP script to calculate and verify the nonce manually:
>> http://pernau.at/kd/sipdigest.php
>>
>> regards
>> Klaus
>>
>> PS: do not use "secret" passwords for testing :-)
>>
>>
>>
>>
>> On 21.08.2012 13:24, David Pati?o wrote:
>>>
>>> Hello,
>>>
>>> I've just test a sip call from an X-Lite client to Ekiga client an
>>> everything has gone ok.
>>> It seems an issue with authentication over WebSockets but I have no
>>> experience in this area so I can not say where the problem is (and
>>> REGISTER auth works fine).
>>>
>>> Any other idea or anything else to check?
>>>
>>> I'm blocked since last week.
>>>
>>> Thanks a lot.
>>> Kind Regards,
>>>
>>> --
>>>        --DAVID--
>>>
>>>
>>>
>>>
>>> 2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com>
>>>>
>>>>
>>>> Hello,
>>>>
>>>> This does look like an issue with authentication rather than WebSockets.
>>>> Have you tried using an ordinary SIP client (for example, Jitsi) with
>>>> your
>>>> Kamailio configuration to verify that the authentication part of
>>>> kamailio.cfg is correct?
>>>>
>>>> Regards,
>>>>
>>>> Peter
>>>>
>>>>
>>>> On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
>>>>
>>>> Hi there,
>>>>
>>>> I'm new on kamailio users list.
>>>> Now I'm developing a client in javascript (using websockets) to SIP
>>>> calls.
>>>>
>>>> I'm using last devel version of kamailio with websocket support. I've
>>>> already get a correct REGISTER transaction but now I'm not able to get
>>>> a correct INVITE transaction. Actually I'm not sure if this a kamailio
>>>> error or a problem with my source code however I'm using the same
>>>> function to get digest info from received nonce both REGISTER and
>>>> INVITE transactions. The INVITE transaction always enters in an
>>>> endless loop with 407 requests.
>>>>
>>>> Please, is there anybody who could help me?
>>>>
>>>> Those are the frames of my code (obtained from chrome browser console):
>>>>
>>>> wsUri: ws://10.1.20.40:80/
>>>> SOCKET CREATED
>>>> SOCKET OPENED
>>>>
>>>> SENT:
>>>> REGISTER sip:10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
>>>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>>>> To: sip:101 at 10.1.20.40
>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>> CSeq: 1 REGISTER
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>>>>          ;reg-id=98
>>>>          ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>>>
>>>>
>>>> RESPONSE:
>>>> SIP/2.0 401 Unauthorized
>>>> Via: SIP/2.0/WS
>>>>
>>>>
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
>>>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>>>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>> CSeq: 1 REGISTER
>>>> WWW-Authenticate: Digest realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> SENT:
>>>> REGISTER sip:10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
>>>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>>>> To: sip:101 at 10.1.20.40
>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>> CSeq: 2 REGISTER
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>>>>          ;reg-id=98
>>>>          ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>>> Authorization: Digest username="101", realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>>> uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
>>>>
>>>>
>>>> RESPONSE:
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/WS
>>>>
>>>>
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
>>>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>>>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>> CSeq: 2 REGISTER
>>>> Contact:
>>>>
>>>> <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> SENT:
>>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 1 INVITE
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>>          ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>>> Content-Type: application/sdp
>>>>
>>>>
>>>> RESPONSE:
>>>> SIP/2.0 407 Proxy Authentication Required
>>>> Via: SIP/2.0/WS
>>>>
>>>>
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 1 INVITE
>>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> SENT:
>>>> ACK sip:102 at 10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 1 ACK
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>>
>>>>
>>>> SENT:
>>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 2 INVITE
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>>          ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>>>> Content-Type: application/sdp
>>>>
>>>>
>>>> RESPONSE:
>>>> SIP/2.0 407 Proxy Authentication Required
>>>> Via: SIP/2.0/WS
>>>>
>>>>
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 2 INVITE
>>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> SENT:
>>>> ACK sip:102 at 10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 2 ACK
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>>
>>>>
>>>> SENT:
>>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>>> Via: SIP/2.0/WS
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 3 INVITE
>>>> Max-Forwards: 70
>>>> Supported: path, outbound, gruu
>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>>          ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>>>> Content-Type: application/sdp
>>>>
>>>>
>>>> RESPONSE:
>>>> SIP/2.0 407 Proxy Authentication Required
>>>> Via: SIP/2.0/WS
>>>>
>>>>
>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
>>>> Call-ID: 9KoKnFdh285k2jg4
>>>> CSeq: 3 INVITE
>>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
>>>> Re"
>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>> Content-Length: 0
>>>>
>>>>
>>>> .......and so on in an endless loop with 407 requests
>>>>
>>>> Thanks a lot
>>>>
>>>> Kind Regards
>>>>
>>>> --
>>>>        --DAVID--
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>> --
>>>> Peter Dunkley
>>>> Technical Director
>>>> Crocodile RCS Ltd
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>


From mkaplan at appcomsci.com  Tue Aug 21 22:52:00 2012
From: mkaplan at appcomsci.com (Kaplan, Michael A)
Date: Tue, 21 Aug 2012 20:52:00 +0000
Subject: [SR-Users] SIP routing configuration howto
Message-ID: <F53A37DAD8610C458EEB8EF9B34D406A0C790C@rrc-ats-exmb1>

Folks,

I'm working on a 5 node network in the following topology:

Host1 - Edge Proxy 1 --- Core Proxy - Edge Proxy 2 - Host2

The proxy machines contain SIP Server installations, the hosts contain user agent software. I am able to make a call end to end between the hosts, but when I look at the traffic in wireshark, the core proxy is actually just forwarding the messages at the IP-layer. I would like actual SIP routing to occur, where a host would send the INVITE to the edge, and the edge has to forward it to the core etc, with these messages actually being processed at the SIP layer for all proxy nodes. I've been trying to use the dynamic routing module, thought I'm not sure this is really right for what I need, simple static routing would likely work as well. Is it necessary for me to rewrite the route{} functions of a proxy script? I'm not sure if I'm approaching this problem the right way. Eventually I would like to have multiple core servers and be able to failover between them, but trying to keep it simple for now.

Thanks,
Mike

--------------------------------
Michael A. Kaplan
Senior Research Scientist
Applied Communication Sciences
732-699-3805 (office)
973-699-6775 (cell)
mkaplan at appcomsci.com


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From serusers at iptel.org  Wed Aug 22 02:38:46 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Tue, 21 Aug 2012 21:38:46 -0300
Subject: [SR-Users] Advertising and instructing company seeking region rep
Message-ID: <50342997.708020@iptel.org>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  Collectively, we focus on different sources of power, new
  green tech, and reusable materials. We are advertising for someone from your
   region to be employed by us. We require: [...] 

Content analysis details:   (11.3 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [189.58.67.119 listed in bb.barracudacentral.org]
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
                            [189.58.67.119 listed in zen.spamhaus.org]
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
 1.3 RCVD_IN_RP_RNBL        RBL: Relay in RNBL,
                            https://senderscore.org/blacklistlookup/
                            [189.58.67.119 listed in bl.score.senderscore.com]
 0.0 RCVD_IN_SORBS_DUL      RBL: SORBS: sent directly from dynamic IP address
                            [189.58.67.119 listed in dnsbl.sorbs.net]
 0.7 SPF_SOFTFAIL           SPF: sender does not match SPF record (softfail)
 1.7 URIBL_DBL_SPAM         Contains an URL listed in the DBL blocklist
                            [URIs: euroinjobs.com]
 1.7 URIBL_BLACK            Contains an URL listed in the URIBL blacklist
                            [URIs: euroinjobs.com]
 0.0 T_FRT_PROFILE2         BODY: ReplaceTags: Profile (2)
-1.9 BAYES_00               BODY: Bayes spam probability is 0 to 1%
                            [score: 0.0001]
 1.6 URIBL_SBL              Contains an URL listed in the SBL blocklist
                            [URIs: euroinjobs.com]
 1.0 RDNS_DYNAMIC           Delivered to internal network by host with
                            dynamic-looking rDNS


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From: <serusers at iptel.org>,
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Size: 1936
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From klaus.mailinglists at pernau.at  Wed Aug 22 08:02:52 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 22 Aug 2012 08:02:52 +0200
Subject: [SR-Users] SIP routing configuration howto
In-Reply-To: <F53A37DAD8610C458EEB8EF9B34D406A0C790C@rrc-ats-exmb1>
References: <F53A37DAD8610C458EEB8EF9B34D406A0C790C@rrc-ats-exmb1>
Message-ID: <5034760C.1030201@pernau.at>

Hi Michael!

Routing at IP layer would only happen if the proxy servers are also 
configured as routers and you have routing entries in the servers.

Depending an what you really need the configuration may differ. Below I 
sketch a very simple scenario with static routing.

On 21.08.2012 22:52, Kaplan, Michael A wrote:
> Folks,
>
> I?m working on a 5 node network in the following topology:
>
> Host1 ? Edge Proxy 1 --- Core Proxy ? Edge Proxy 2 ? Host2

Ok. Lets give them names and addresses.

Host1 ? Edge Proxy 1 --- Core Proxy ? Edge Proxy 2 ? Host2
           2.2.2.2                     4.4.4.4
1 at 1.1.1.1                 3.3.3.3                   5 at 5.5.5.5

UAC1 has the SIP URI sip:1 at 1.1.1.1, UAC2 has the SIP URI sip:5 at 5.5.5.5.

UAC1 calls UAC2 with static routing. UAC1 is configured to use 2.2.2.2 
as outbound proxy. Thus, UAC1 sends to Edge Proxy1 (ep1):

INVITE sip:5 at 5.5.5.5
Via: SIP/2.0/1.1.1.1:5060....

ep1 is configured to send everything, which does not come from the core 
proxy (cp), to the core proxy:

   if ($si != "3.3.3.3") {
     # forward to cp without changing the request URI
     $du = "sip:3.3.3.3";
   }
   t_relay();
   exit;

Thus, ep1 sends to cp:

INVITE sip:5 at 5.5.5.5
Via: SIP/2.0/2.2.2.2:5060....
Via: SIP/2.0/1.1.1.1:5060....

The cp has a static routing and knows which ep is responsible for a 
respective user.

   if ($ru == "sip:1 at 1.1.1.1") {
     # forward to ep without changing the request URI
     $du = "sip:2.2.2.2";
   } else if ($ru == "sip:5 at 5.5.5.5") {
     # forward to ep without changing the request URI
     $du = "sip:4.4.4.4";
   } else {
     sl_send_reply("404","Unknown User");
     exit;
   }
   t_relay();
   exit;

Thus, cp sends to ep2:

INVITE sip:5 at 5.5.5.5
Via: SIP/2.0/3.3.3.3:5060....
Via: SIP/2.0/2.2.2.2:5060....
Via: SIP/2.0/1.1.1.1:5060....


ep2 uses the same config as ep1 above and you are done (with this simple 
scenario). In dialog can be routed directly end-2-end or via the proxys 
when using record_route() and loose_route.

Note: This simple routing config does not give you any security (if this 
is relevant in your setup).

PS: With registration things get more complex and you should really 
understand the default configuration file and start playing around with 
1 proxy, or 1 proxy and 1 edge proxy, before building such complex networks.

PSS: If the proxyies are multihomed you should set mhomed=1

regards
Klaus





>
> The proxy machines contain SIP Server installations, the hosts contain
> user agent software. I am able to make a call end to end between the
> hosts, but when I look at the traffic in wireshark, the core proxy is
> actually just forwarding the messages at the IP-layer. I would like
> actual SIP routing to occur, where a host would send the INVITE to the
> edge, and the edge has to forward it to the core etc, with these
> messages actually being processed at the SIP layer for all proxy nodes.
> I?ve been trying to use the dynamic routing module, thought I?m not sure
> this is really right for what I need, simple static routing would likely
> work as well. Is it necessary for me to rewrite the route{} functions of
> a proxy script? I?m not sure if I?m approaching this problem the right
> way. Eventually I would like to have multiple core servers and be able
> to failover between them, but trying to keep it simple for now.
>
> Thanks,
>
> Mike
>
> --------------------------------
>
> Michael A. Kaplan
>
> Senior Research Scientist
>
> Applied Communication Sciences
>
> 732-699-3805 (office)
>
> 973-699-6775 (cell)
>
> mkaplan at appcomsci.com
>
> **
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From david.patino at quobis.com  Wed Aug 22 09:57:51 2012
From: david.patino at quobis.com (=?ISO-8859-1?Q?David_Pati=F1o?=)
Date: Wed, 22 Aug 2012 09:57:51 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <5033C895.9020605@pernau.at>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
	<5033864C.1070904@pernau.at>
	<CABmM4BCcyDHhd69B289JaMS7nC6Zs+_0CmNdri-veLZojKiC9g@mail.gmail.com>
	<5033C895.9020605@pernau.at>
Message-ID: <CABmM4BCb7qGWOvH57rOE-4YaWgABOdJ2wvLfNQd4aUppbUBv6A@mail.gmail.com>

Hi there,

Awesome Klaus! A lot of thanks!
You're absolutely right, I've made a mistake calculating the response
because of I've been using the wrong URI until now.

On the other hand, during a REGISTER dialog I've been using
sip:10.1.20.40 as URI parameter to calculate 401 response and it's
still being registered properly, I've realized about that due to
INVITE response errors, is it correct?

Thank you very much.
Kind Regards,

--
     --DAVID--



2012/8/21 Klaus Darilion <klaus.mailinglists at pernau.at>:
> When you send the INVITE, then the RURI is not sip:10.1.20.40 but
> sip:102 at 10.1.20.40 - it includes the user part.
>
> I see that you have it correct in the Proxy-Authorization header of the
> INVITE, but maybe you calculated the response with the wrong URI.
>
> Anyway, taking a look at your INVITE from your previous email:
>
>
> INVITE sip:102 at 10.1.20.40 SIP/2.0
> Via: SIP/2.0/WS
> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
> To: sip:102 at 10.1.20.40
> Call-ID: 9KoKnFdh285k2jg4
> CSeq: 3 INVITE
> Max-Forwards: 70
> Supported: path, outbound, gruu
> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>         ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
> Content-Type: application/sdp
>
> This gives as input to my script:
> username: 101
> password: 101 (hope this is correct)
> realm: 10.1.20.40
> nonce: UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re
> uri: sip:102 at 10.1.20.40
> method: INVITE
>
> --> my script calculates 4a6dab565fa5f1e39e13960c2f323eab but the INVITE
> contains f4bbe128abf3acb3a4e11882749d8798. So, your stack seems to be buggy.
>
> regards
> klaus
>
>
>
>
>
> On 21.08.2012 16:05, David Pati?o wrote:
>>
>> Hi,
>>
>> Awesome tool!
>>
>> I've been checking my info with the output of your script and the
>> output is the same. Maybe the auth problem is not here or maybe I've
>> made mistakes. Could you please tell my if the data I'm using is
>> correct?
>>
>> For example, I'm user 101 and I want to call to user 102. So I've used
>> the following values to create the INVITE response to 407 message:
>>
>> User: 101
>> Pass: 101
>> realm: 10.1.20.40
>> nonce:"UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv"
>> uri: sip:10.1.20.40
>>
>> HA1 params: 101:10.1.20.40:101
>> HA1: 1d578520f175df632f0850415e603029
>> HA2 params: INVITE:sip:10.1.20.40
>> HA2: 793dd6fa2e181e25a226cc09efc6dc2c
>> Sol params:
>> 1d578520f175df632f0850415e603029:UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv:793dd6fa2e181e25a226cc09efc6dc2c
>> Sol: 362206a07a3782c6e2557699abe5bf7b
>>
>> Are those values correct?
>>
>> Thank you very much.
>> Kind Regards,
>>
>> --
>>       --DAVID--
>>
>>
>>
>> 2012/8/21 Klaus Darilion <klaus.mailinglists at pernau.at>:
>>>
>>> I just hacked a small PHP script to calculate and verify the nonce
>>> manually:
>>> http://pernau.at/kd/sipdigest.php
>>>
>>> regards
>>> Klaus
>>>
>>> PS: do not use "secret" passwords for testing :-)
>>>
>>>
>>>
>>>
>>> On 21.08.2012 13:24, David Pati?o wrote:
>>>>
>>>>
>>>> Hello,
>>>>
>>>> I've just test a sip call from an X-Lite client to Ekiga client an
>>>> everything has gone ok.
>>>> It seems an issue with authentication over WebSockets but I have no
>>>> experience in this area so I can not say where the problem is (and
>>>> REGISTER auth works fine).
>>>>
>>>> Any other idea or anything else to check?
>>>>
>>>> I'm blocked since last week.
>>>>
>>>> Thanks a lot.
>>>> Kind Regards,
>>>>
>>>> --
>>>>        --DAVID--
>>>>
>>>>
>>>>
>>>>
>>>> 2012/8/21 Peter Dunkley <peter.dunkley at crocodile-rcs.com>
>>>>>
>>>>>
>>>>>
>>>>> Hello,
>>>>>
>>>>> This does look like an issue with authentication rather than
>>>>> WebSockets.
>>>>> Have you tried using an ordinary SIP client (for example, Jitsi) with
>>>>> your
>>>>> Kamailio configuration to verify that the authentication part of
>>>>> kamailio.cfg is correct?
>>>>>
>>>>> Regards,
>>>>>
>>>>> Peter
>>>>>
>>>>>
>>>>> On Tue, 2012-08-21 at 10:59 +0200, David Pati?o wrote:
>>>>>
>>>>> Hi there,
>>>>>
>>>>> I'm new on kamailio users list.
>>>>> Now I'm developing a client in javascript (using websockets) to SIP
>>>>> calls.
>>>>>
>>>>> I'm using last devel version of kamailio with websocket support. I've
>>>>> already get a correct REGISTER transaction but now I'm not able to get
>>>>> a correct INVITE transaction. Actually I'm not sure if this a kamailio
>>>>> error or a problem with my source code however I'm using the same
>>>>> function to get digest info from received nonce both REGISTER and
>>>>> INVITE transactions. The INVITE transaction always enters in an
>>>>> endless loop with 407 requests.
>>>>>
>>>>> Please, is there anybody who could help me?
>>>>>
>>>>> Those are the frames of my code (obtained from chrome browser console):
>>>>>
>>>>> wsUri: ws://10.1.20.40:80/
>>>>> SOCKET CREATED
>>>>> SOCKET OPENED
>>>>>
>>>>> SENT:
>>>>> REGISTER sip:10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg
>>>>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>>>>> To: sip:101 at 10.1.20.40
>>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>>> CSeq: 1 REGISTER
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>>>>>          ;reg-id=98
>>>>>          ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>>>>
>>>>>
>>>>> RESPONSE:
>>>>> SIP/2.0 401 Unauthorized
>>>>> Via: SIP/2.0/WS
>>>>>
>>>>>
>>>>>
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229
>>>>> From: sip:101 at 10.1.20.40;tag=Me0arqDG6K8o7l32oJgR
>>>>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c
>>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>>> CSeq: 1 REGISTER
>>>>> WWW-Authenticate: Digest realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> SENT:
>>>>> REGISTER sip:10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037
>>>>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>>>>> To: sip:101 at 10.1.20.40
>>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>>> CSeq: 2 REGISTER
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws>
>>>>>          ;reg-id=98
>>>>>          ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>"
>>>>> Authorization: Digest username="101", realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>>>> uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da"
>>>>>
>>>>>
>>>>> RESPONSE:
>>>>> SIP/2.0 200 OK
>>>>> Via: SIP/2.0/WS
>>>>>
>>>>>
>>>>>
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229
>>>>> From: sip:101 at 10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612
>>>>> To: sip:101 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4
>>>>> Call-ID: 0EQSBkoG9RGgfg3E
>>>>> CSeq: 2 REGISTER
>>>>> Contact:
>>>>>
>>>>>
>>>>> <sip:101 at KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101 at 10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173 at 10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98
>>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> SENT:
>>>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 1 INVITE
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>>>          ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>>>> Content-Type: application/sdp
>>>>>
>>>>>
>>>>> RESPONSE:
>>>>> SIP/2.0 407 Proxy Authentication Required
>>>>> Via: SIP/2.0/WS
>>>>>
>>>>>
>>>>>
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 1 INVITE
>>>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> SENT:
>>>>> ACK sip:102 at 10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 1 ACK
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>>
>>>>>
>>>>> SENT:
>>>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 2 INVITE
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>>>          ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>>>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>>>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>>>>> Content-Type: application/sdp
>>>>>
>>>>>
>>>>> RESPONSE:
>>>>> SIP/2.0 407 Proxy Authentication Required
>>>>> Via: SIP/2.0/WS
>>>>>
>>>>>
>>>>>
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 2 INVITE
>>>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re"
>>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> SENT:
>>>>> ACK sip:102 at 10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 2 ACK
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>>
>>>>>
>>>>> SENT:
>>>>> INVITE sip:102 at 10.1.20.40 SIP/2.0
>>>>> Via: SIP/2.0/WS
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 3 INVITE
>>>>> Max-Forwards: 70
>>>>> Supported: path, outbound, gruu
>>>>> Contact: <sip:101 at KMkjMf37lVeM.invalid;transport=ws
>>>>>          ;gr=urn:uuid:beb-68a9-6396e5f4;ob>
>>>>> Proxy-Authorization: Digest username="101", realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="",
>>>>> uri="sip:102 at 10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798"
>>>>> Content-Type: application/sdp
>>>>>
>>>>>
>>>>> RESPONSE:
>>>>> SIP/2.0 407 Proxy Authentication Required
>>>>> Via: SIP/2.0/WS
>>>>>
>>>>>
>>>>>
>>>>> KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
>>>>> From: sip:101 at 10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
>>>>> To: sip:102 at 10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
>>>>> Call-ID: 9KoKnFdh285k2jg4
>>>>> CSeq: 3 INVITE
>>>>> Proxy-Authenticate: Digest realm="10.1.20.40",
>>>>> nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4
>>>>> Re"
>>>>> Server: kamailio (3.4.0-dev3 (i386/linux))
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>> .......and so on in an endless loop with 407 requests
>>>>>
>>>>> Thanks a lot
>>>>>
>>>>> Kind Regards
>>>>>
>>>>> --
>>>>>        --DAVID--
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>> --
>>>>> Peter Dunkley
>>>>> Technical Director
>>>>> Crocodile RCS Ltd
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>


From patrice.bodeven at orange.com  Wed Aug 22 10:51:26 2012
From: patrice.bodeven at orange.com (patrice.bodeven at orange.com)
Date: Wed, 22 Aug 2012 08:51:26 +0000
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
Message-ID: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>

Hello,

I am working on Kamailio 3.2.2. There is no traffic, only functional test done.

Based on the SIP Client used until now (Xlite), the INVITE is systematically authenticated by 407 as there is no Proxy-Authorization in the initial INVITE. Expected/normal behavior.

But when using an internal SIP client, the initial INVITE is generated with a Proxy-Authorization containing the nonce value used by the successful registration.
In Authorization of Register (the answer is 200OK)
Digest username="+33296488922 at ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:ims.v0.pftest.net", response="4ed51b8e62cf769b19f8c4771e068ad3", cnonce="06440000000529330fe1", qop=auth, nc=00000011

In Proxy-Authorization of Invite
Digest username="+33296488922 at ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:0157361149 at ims.v0.pftest.net", response="b247052d6d5b37e0b6677c816390cb77", cnonce="06440000000529330fe1", qop=auth, nc=00000012

As the nonce expire is high (see nonce_expire), I don't understand why Kamailio requests the Authentication of the INVITE by 407.
407 with Proxy-Authenticate: Digest realm="ims.v0.pftest.net", nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true

Does it mean that the REGISTER and INVITE are using different Nonce value ?

==> Information on my configuration file.

modparam("auth_db", "db_url", DBURLAUTH)
modparam("auth", "auth_checks_register", 6) # callid and From TAG
modparam("auth", "qop", "auth")    # enable qop=auth
modparam("auth", "nonce_count", 1) # enable nonce_count support
modparam("auth", "nonce_expire", 21600)   # Set nonce_expire to 6 hours

For register, I do
                 if (!www_authorize("$ru", "subscriber"))
                {
                   if($? == -2 || $? == -3) {
                      sl_send_reply("403","Forbidden auth ID");
                     exit;
                   }
                   else {
                        www_challenge("$ru", "17");
                        exit;
                   }
                }

For invite, I do
           if (!proxy_authorize("$fd", "subscriber"))
           {
               if($? == -2 || $? == -3) {
                  sl_send_reply("403","Forbidden auth ID");
                  exit;
               }
              else {
                   proxy_challenge("$fd", "17");
                   xlog("L_INFO","INVITE challenged by 407 to :$di:$dp \n");
                   exit;
                }
           }

Followed by consume_credentials();

Thanks in advance for your support.
Patrice B


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From phillman25 at gmail.com  Wed Aug 22 12:02:57 2012
From: phillman25 at gmail.com (phillman25)
Date: Wed, 22 Aug 2012 13:02:57 +0300
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
Message-ID: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>

Dear List

I am trying to run the below command from a REMOTE server where i use Cacti
to graph all my data. I am trying to graph on a per trunk basis.

kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'

This command yields an output on the local server.

Could someone perhaps point me in the right direction?


Thanks!
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From jh at tutpro.com  Wed Aug 22 12:18:50 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Wed, 22 Aug 2012 13:18:50 +0300
Subject: [SR-Users] new feature: option to remove contacts based on sip nat
	keepalives
Message-ID: <20532.45578.472915.54462@siika.tutpro.com>

daniel,

you wrote:

  This feature should be useful in mobile networks, to avoid stacking lot 
  of invalid contacts which could result in lot of branches causing many 
  retransmissions.

why would a sip ua that runs in a mobile network device be configured to
use udp instead of tcp?

poor battery life seems to be a big problem in so called smart phones.
wouldn't this kind of active pinging make it even worse?

-- juha


From klaus.mailinglists at pernau.at  Wed Aug 22 14:03:20 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 22 Aug 2012 14:03:20 +0200
Subject: [SR-Users] Websocket INVITE 407 error
In-Reply-To: <CABmM4BCb7qGWOvH57rOE-4YaWgABOdJ2wvLfNQd4aUppbUBv6A@mail.gmail.com>
References: <CABmM4BAD6yJPeksk1O+ZxzYMuSy5EM32iQjtwjWx-K+L3TbwaQ@mail.gmail.com>
	<1345547090.13632.2.camel@pd-notebook-linux.croc.internal>
	<CABmM4BDFKcPUmWYgoesiRMrOKc886H4S0YUYC406miVNtJs9ng@mail.gmail.com>
	<5033864C.1070904@pernau.at>
	<CABmM4BCcyDHhd69B289JaMS7nC6Zs+_0CmNdri-veLZojKiC9g@mail.gmail.com>
	<5033C895.9020605@pernau.at>
	<CABmM4BCb7qGWOvH57rOE-4YaWgABOdJ2wvLfNQd4aUppbUBv6A@mail.gmail.com>
Message-ID: <5034CA88.3080909@pernau.at>



On 22.08.2012 09:57, David Pati?o wrote:
> Hi there,
>
> Awesome Klaus! A lot of thanks!
> You're absolutely right, I've made a mistake calculating the response
> because of I've been using the wrong URI until now.
>
> On the other hand, during a REGISTER dialog I've been using
> sip:10.1.20.40 as URI parameter to calculate 401 response and it's
> still being registered properly, I've realized about that due to
> INVITE response errors, is it correct?

Yes thats correct. you have to use the URI from the request line. During 
REGISTER it only contains the hostname, no userpart.

regards
Klaus


From miconda at gmail.com  Wed Aug 22 14:17:14 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 22 Aug 2012 14:17:14 +0200
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
Message-ID: <5034CDCA.3050604@gmail.com>

Hello,

On 8/22/12 10:51 AM, patrice.bodeven at orange.com wrote:
>
> Hello,
>
> I am working on Kamailio 3.2.2. There is no traffic, only functional 
> test done.
>
> Based on the SIP Client used until now (Xlite), the INVITE is 
> systematically authenticated by 407 as there is no Proxy-Authorization 
> in the initial INVITE. Expected/normal behavior.
>
> But when using an internal SIP client, the initial INVITE is generated 
> with a Proxy-Authorization containing the nonce value used by the 
> successful registration.
>
> _In Authorization of Register (the answer is 200OK)_
>
> Digest username="+33296488922 at ims.v0.pftest.net", 
> realm="sip:ims.v0.pftest.net", 
> nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", 
> uri="sip:ims.v0.pftest.net", 
> response="4ed51b8e62cf769b19f8c4771e068ad3", 
> cnonce="06440000000529330fe1", qop=auth, nc=00000011
>
> _In Proxy-Authorization of Invite _
>
> Digest username="+33296488922 at ims.v0.pftest.net", 
> realm="sip:ims.v0.pftest.net", 
> nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", 
> uri="sip:0157361149 at ims.v0.pftest.net", 
> response="b247052d6d5b37e0b6677c816390cb77", 
> cnonce="06440000000529330fe1", qop=auth, nc=00000012
>
> As the nonce expire is high (see nonce_expire), I don't understand why 
> Kamailio requests the Authentication of the INVITE by 407.
>
> _407 with Proxy-Authenticate_: Digest realm="ims.v0.pftest.net", 
> nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true
>
> *Does it mean that the REGISTER and INVITE are using different Nonce 
> value ? *
>
> ?Information on my configuration file.
>
> modparam("auth_db", "db_url", DBURLAUTH)
>
> modparam("auth", "auth_checks_register", 6) # callid and From TAG
>
> modparam("auth", "qop", "auth")    # enable qop=auth
>
> modparam("auth", "nonce_count", 1) # enable nonce_count support
>
> modparam("auth", "nonce_expire", 21600)   # Set nonce_expire to 6 hours
>
> For register, I do
>
>                  if (!www_authorize("$ru", "subscriber"))
>
>                 {
>
>                    if($? == -2 || $? == -3) {
>
>             sl_send_reply("403","Forbidden auth ID");
>
>            exit;
>
>                    }
>
>                    else {
>
>         www_challenge("$ru", "17");
>
>     exit;
>
>                    }
>
>                 }
>
> For invite, I do
>
>            if (!proxy_authorize("$fd", "subscriber"))
>
>            {
>
>                if($? == -2 || $? == -3) {
>
>     sl_send_reply("403","Forbidden auth ID");
>
>                   exit;
>
>                }
>
>               else {
>
> proxy_challenge("$fd", "17");
>
>    xlog("L_INFO","INVITE challenged by 407 to :$di:$dp \n");
>
>                    exit;
>
>                 }
>
>            }
>
> Followed by consume_credentials();
>
> Thanks in advance for your support.
>
set debug=3 in your configuration file and watch the log messages. You 
should get some leads about why the authentication is not succeeding.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Wed Aug 22 14:18:50 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 22 Aug 2012 14:18:50 +0200
Subject: [SR-Users] new feature: option to remove contacts based on sip
 nat keepalives
In-Reply-To: <20532.45578.472915.54462@siika.tutpro.com>
References: <20532.45578.472915.54462@siika.tutpro.com>
Message-ID: <5034CE2A.6020403@gmail.com>

Hello,

On 8/22/12 12:18 PM, Juha Heinanen wrote:
> daniel,
>
> you wrote:
>
>    This feature should be useful in mobile networks, to avoid stacking lot
>    of invalid contacts which could result in lot of branches causing many
>    retransmissions.
>
> why would a sip ua that runs in a mobile network device be configured to
> use udp instead of tcp?
>
> poor battery life seems to be a big problem in so called smart phones.
> wouldn't this kind of active pinging make it even worse?
yes, pinging could make it worse. Personally I recommend TLS for mobile 
networks, to protect privacy of instant messaging, specially if there is 
going to be usage over public wi-fi spots. That also adds more 
processing to encrypt/decrypt, but it is the price of better security.

However, there are lot of sip softphones apps that still do only udp, in 
many cases resulting in lots of contacts in location table. Here can 
help also the option of saving one contact per user (via parameter of 
save() function), if it is known that the user should have only one device.

All over, the new functionality is optional and off by default. It is 
the decision of admin to enable or not, for mobile or fixed networks.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From vijay.thakur at loopmethods.com  Wed Aug 22 14:26:47 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Wed, 22 Aug 2012 17:56:47 +0530
Subject: [SR-Users] Kamailio with asterisk for outbound calls
Message-ID: <5034D007.6040608@loopmethods.com>

Hi All Kamailio Experts,

I have configured Kamailio (kamailio 3.1.5) as media server. All things 
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound 
Calls. For this purpose i have followed the web page : 
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. 
In this page, some points are not clear for me , as given below:

(1) In case you use *sipregs* you have to create a record for each 
extension where to set the 'name' to value of 'name' from *sipusers*. 
The rest is populated by Asterisk from registrations.

(2) Be sure you configure Asterisk *to not authenticate* SIP requests 
coming from Kamailio.

I am not sure that my local users chat is working through kamailio or 
asterisk, who is used for authorization.
Any specific Web page to correct the issue will highly appreciated 
according to my scenario.

Kindly guide me. Thanks in advance.

-- 
Best Regards,

Vijay Thakur
(Assistant Manager - Networks)
Mobile   : +91 8744018065
Mail     : vijay.thakur at loopmethods.com

Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web: www.loopmethods.com

LOOP Disclaimer -------------------------------------------------------------------------------------------------
This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited.
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From oyvind.kolbu at usit.uio.no  Wed Aug 22 15:32:40 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Wed, 22 Aug 2012 15:32:40 +0200
Subject: [SR-Users] db_cluster together with the registrar module = signal 11
Message-ID: <20120822133240.GF16374@putsch.kolbu.ws>

Hi,

I upgraded our system to Kamailio 3.3 to take advantage of the new
db_cluster module. Currently we have two registration/location servers
each with their own sql-server. Authenticated REGISTER messages are
forwarded to the other server, which uses the 0x02 flag to save() in
order to store it without generating an extra reply. This works fine.

My goal was to use db_cluster to update both databases at once with save()
and use the failover mode when reading with lookup(). Have anyone tested
a similar setup?

Anyhow, here is how I tried to do it:

modparam("db_cluster","connection","voip1_data2=>postgres://user:pass at hostname1/voipt1_data2")
modparam("db_cluster","connection","voip2_data2=>postgres://user:pass at hostname2/voipt2_data2")
modparam("db_cluster", "cluster", "data2=>voip1_data2=8s7p;voip2_data2=9s7p")

modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url", "cluster://data2")

But I get:

    : : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33
    : ALERT: <core> [main.c:785]: child process 28275 exited by a signal 11
    : ALERT: <core> [main.c:788]: core was generated
    : INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD

Same with db_mode = 1 or 3, and either
    modparam("db_cluster", "cluster", "data2=>voip1_data2=8s9p")
or
    modparam("db_cluster", "cluster", "data2=>voip2_data2=8s9p").

No difference between newly booted phones or phones reregistering.

Backtrace from gdb:

(gdb) bt
#0  0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334
#1  0x00fbb1cc in db_insert_ucontact (_c=0xb5d828d8) at ucontact.c:565
#2  0x00fcfc5d in insert_ucontact (_r=0xfd3a00, _contact=0xb7d3ce90, _ci=0xc5cc80, _c=0xbf9acb84) at urecord.c:478
#3  0x00c55bb0 in add_contacts (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:493
#4  save (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:864
#5  0x00c4dd43 in w_save2 (_m=0xb7d05e40, _d=0xb5d63060 "00?", _cflags=0x0) at reg_mod.c:407
#6  0x080586ad in do_action (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1157
#7  0x0805fe5c in run_actions (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1644
#8  0x080604e1 in run_actions_safe (h=0xbf9adcb8, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1708
#9  0x080f96b7 in rval_get_int (h=0xbf9adcb8, msg=0xbf9ac954, i=0xbf9ad268, rv=0xb7c406cc, cache=0x0) at rvalue.c:920
#10 0x080fcf38 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad268, rve=0xb7c406c8) at rvalue.c:1914
#11 0x080fd009 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad544, rve=0xb7c40a70) at rvalue.c:1922
#12 0x080585fe in do_action (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1121
#13 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1644
#14 0x08058657 in do_action (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1136
#15 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1644
#16 0x0805853d in do_action (h=0xbf9adcb8, a=0xb7bd64d8, msg=0xb7d05e40) at action.c:761
#17 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7bc7128, msg=0xb7d05e40) at action.c:1644
#18 0x08060464 in run_top_route (a=0xb7bc7128, msg=0xb7d05e40, c=0x0) at action.c:1729
#19 0x080e11e4 in receive_msg (buf=0x82c4f20 "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619 at hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"...,
    len=809, rcv_info=0xbf9ade98) at receive.c:209
#20 0x08177f7b in udp_rcv_loop () at udp_server.c:544
#21 0x080b0426 in main_loop () at main.c:1633
#22 0x080b3d02 in main (argc=17, argv=0xbf9ae174) at main.c:2546

Core dumps and debug logs (debug=3) are available.

-- 
?yvind Kolbu
University of Oslo


From oyvind.kolbu at usit.uio.no  Wed Aug 22 15:37:43 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Wed, 22 Aug 2012 15:37:43 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <20120822133240.GF16374@putsch.kolbu.ws>
References: <20120822133240.GF16374@putsch.kolbu.ws>
Message-ID: <20120822133743.GG16374@putsch.kolbu.ws>

On 2012-08-22 at 15:32, ?yvind Kolbu wrote:
[...]

Some more details:

# kamailio -V
version: kamailio 3.3.1 (i386/linux) aae4e4
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: aae4e4
compiled on 16:30:51 Aug 21 2012 with gcc 4.1.2

Kamailio 3.3 branch as of git fb51d3815292c27245b26d1f4a4f6c9b190f9200.

-- 
?yvind Kolbu


From miconda at gmail.com  Wed Aug 22 16:01:00 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 22 Aug 2012 16:01:00 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120822133743.GG16374@putsch.kolbu.ws>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws>
Message-ID: <5034E61C.60600@gmail.com>

Hello,

can you send the output of 'bt full' from gdb?

Simple bt shows only the line where a define is executed, hopefully bt 
full will get the values from the inside defines to spot the problem.

Also, can you give it a quick try with 9p instead of 7p for both 
connections?

Cheers,
Daniel


On 8/22/12 3:37 PM, ?yvind Kolbu wrote:
> On 2012-08-22 at 15:32, ?yvind Kolbu wrote:
> [...]
>
> Some more details:
>
> # kamailio -V
> version: kamailio 3.3.1 (i386/linux) aae4e4
> flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> id: aae4e4
> compiled on 16:30:51 Aug 21 2012 with gcc 4.1.2
>
> Kamailio 3.3 branch as of git fb51d3815292c27245b26d1f4a4f6c9b190f9200.
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From bligthart at btlnet.co.uk  Wed Aug 22 17:34:17 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Wed, 22 Aug 2012 15:34:17 +0000
Subject: [SR-Users] Kamailio Crash using Cassandra 1.1.2 during SIP
	unregister
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11D150@DB3PRD0510MB380.eurprd05.prod.outlook.com>


Hi,

Kamailio crash using Cassandra 1.1.2 during SIP unregister (registering works fine)

The version of Kamailio is the latest GIT version:

version: kamailio 3.4.0-dev3 (x86_64/linux) 10327c
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 10327c
compiled on 14:53:11 Aug 22 2012 with gcc 4.6.3

BT of Core :

Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `kamailio -ddddd'.
Program terminated with signal 11, Segmentation fault.
#0  fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527
527             if(unlikely(f->u.nxt_free!=NULL)) {
(gdb) bt
#0  fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527
#1  0x00007fa55d35ac53 in db_free_columns (_r=0x7fa55db38c30) at db_res.c:86
#2  0x00007fa55d83f87d in db_cassa_query (_h=<optimized out>, _k=0x7fa55adc6aac, _op=<optimized out>, _v=<optimized out>, _c=0x7fff72ef0e30, _n=<optimized out>,
    _nc=16, _o=0x7fa55afca990, _r=0x7fff72ef0f10) at dbcassa_base.cpp:659
#3  0x00007fa55adb58eb in db_load_urecord (_c=0x7fa55db218d0, _d=0x7fa5576eee38, _aor=0x7fff72ef11b0) at udomain.c:592
#4  0x00007fa55adb93cd in get_urecord (_d=<optimized out>, _aor=0x7fff72ef11b0, _r=0x7fff72ef1200) at udomain.c:1017
#5  0x00007fa55ab91698 in lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38, _uri=<optimized out>) at lookup.c:156
#6  0x00007fa55ab957f2 in w_lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38 "\350\355nW\245\177", _uri=<optimized out>) at reg_mod.c:437
#7  0x000000000041c624 in do_action (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1153
#8  0x000000000041b493 in run_actions (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1646
#9  0x0000000000423a24 in run_actions_safe (h=<optimized out>, a=<optimized out>, msg=<optimized out>) at action.c:1710
#10 0x00000000004d79a0 in rval_get_int (h=0x7fff72ef2790, msg=<optimized out>, i=0x7fff72ef1c30, rv=<optimized out>, cache=<optimized out>) at rvalue.c:920
#11 0x00000000004dc82c in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef1c30, rve=0x7fa55db23218) at rvalue.c:1914
#12 0x00000000004dc8f9 in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef20d0, rve=0x7fa55db238e8) at rvalue.c:1922
#13 0x000000000041cb66 in do_action (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1123
#14 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1646
#15 0x000000000041c7e1 in do_action (h=0x7fff72ef2790, a=0x7fa55db15b38, msg=0x7fa55db40c48) at action.c:763
#16 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db118c8, msg=0x7fa55db40c48) at action.c:1646
#17 0x0000000000423ab0 in run_top_route (a=0x7fa55db118c8, msg=0x7fa55db40c48, c=<optimized out>) at action.c:1731
#18 0x00000000004b39ee in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:209
#19 0x000000000054de1a in udp_rcv_loop () at udp_server.c:544
#20 0x0000000000478879 in main_loop () at main.c:1635
#21 0x000000000041b002 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2548

Thanks,

Bou
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From carlos at nbtbizcapital.com  Wed Aug 22 18:01:28 2012
From: carlos at nbtbizcapital.com (Carlos Cruz)
Date: Wed, 22 Aug 2012 12:01:28 -0400
Subject: [SR-Users] interfacing with kamailio
Message-ID: <01c801cd807f$63c88520$2b598f60$@nbtbizcapital.com>

I'm just starting to evaluate Kamailio to be used as a front end to a couple
of Asterisk Servers.

 

Currently I have an Adobe Flex application that interfaces with Asterisk via
AMI using a socket connection. I searched on how to interface with Kamailio
but wasn't able to find anything that helped me.  I would appreciate it if
someone could point me to where I can find information on how this can be
done using Kamailio. For example I would like my Flex Application to connect
to Kamailio to know the status of the SIP channels.

 

Thanks for any help!!!

 

Carlos

 

 

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From oej at edvina.net  Wed Aug 22 18:24:19 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Wed, 22 Aug 2012 18:24:19 +0200
Subject: [SR-Users] interfacing with kamailio
In-Reply-To: <01c801cd807f$63c88520$2b598f60$@nbtbizcapital.com>
References: <01c801cd807f$63c88520$2b598f60$@nbtbizcapital.com>
Message-ID: <5E42AE8C-CDCF-4128-A14A-B0D83A695F44@edvina.net>


22 aug 2012 kl. 18:01 skrev Carlos Cruz:

> I'm just starting to evaluate Kamailio to be used as a front end to a couple of Asterisk Servers.
>  
> Currently I have an Adobe Flex application that interfaces with Asterisk via AMI using a socket connection. I searched on how to interface with Kamailio but wasn't able to find anything that helped me.  I would appreciate it if someone could point me to where I can find information on how this can be done using Kamailio. For example I would like my Flex Application to connect to Kamailio to know the status of the SIP channels.

By default, kamailio doesn't have any knowledge about "channel". A SIP proxy is transaction stageful - in telephony terminology, it knows a call setup or a teardown, but doesn't have any relationship between them - the idea of a "call" is lost.

You can make Kamailio call stateful by using the dialog module, but on the other side - you don't need it since Asterisk is very call stateful. In fact, it makes your platform less redundant to have call states everywhere.

The interface you are looking for is either the MI interface or the RPC interface. We have two interfaces that mostly provides the same functionality. Every module has different functions. It is a simple query-answer interface and has not got the "events" that you get from Asterisk.

I hope this helps you.

/Olle

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From oyvind.kolbu at usit.uio.no  Wed Aug 22 18:34:34 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Wed, 22 Aug 2012 18:34:34 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <5034E61C.60600@gmail.com>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
Message-ID: <20120822163434.GH16374@putsch.kolbu.ws>

On 2012-08-22 at 16:01, Daniel-Constantin Mierla wrote:
> Hello,
> 
> can you send the output of 'bt full' from gdb?

#0  0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334
        ret = -1
        i = 7
        k = -1212405280
        rc = 8
        rok = 0
        j = 10624544
        cls = 0xb7bc29e0
#1  0x00fbb1cc in db_insert_ucontact (_c=0xb5d828d8) at ucontact.c:565
        keys = {0xfd3400, 0xfd3410, 0xfd3418, 0xfd3420, 0xfd3428, 0xfd3430, 0xfd3438, 0xfd3440, 0xfd3448, 0xfd3450, 0xfd3458, 0xfd3460, 0xfd3468, 0xfd3480, 0xfd33f8, 0xfd3470, 0xfd3478, 0xfd3408}
        vals = {{type = DB1_STR, nul = 0, free = 174331528, val = {int_val = 12962272, ll_val = 30077733344, double_val = 1.4860374750044156e-313, time_val = 12962272, string_val = 0xc5c9e0 "2549619 at hometest.voip.uio.no", str_val = {
                s = 0xc5c9e0 "2549619 at hometest.voip.uio.no", len = 7}, blob_val = {s = 0xc5c9e0 "2549619 at hometest.voip.uio.no", len = 7}, bitmap_val = 12962272}}, {type = DB1_STR, nul = 0, free = 10809332, val = {int_val = -1244124848, ll_val = 114719992144,
              double_val = 5.667920700952809e-313, time_val = -1244124848, string_val = 0xb5d82950 "sip:2549619 at 193.157.149.56", str_val = {s = 0xb5d82950 "sip:2549619 at 193.157.149.56", len = 26}, blob_val = {s = 0xb5d82950 "sip:2549619 at 193.157.149.56", len = 26},
              bitmap_val = 3050842448}}, {type = DB1_DATETIME, nul = 0, free = 173661600, val = {int_val = 1345641342, ll_val = 1345641342, double_val = 6.6483515870591172e-315, time_val = 1345641342, string_val = 0x5034db7e <Address 0x5034db7e out of bounds>,
              str_val = {s = 0x5034db7e <Address 0x5034db7e out of bounds>, len = 0}, blob_val = {s = 0x5034db7e <Address 0x5034db7e out of bounds>, len = 0}, bitmap_val = 1345641342}}, {type = DB1_DOUBLE, nul = 0, free = 173661600, val = {int_val = 0,
              ll_val = -4616189618054758400, double_val = -1, time_val = 0, string_val = 0x0, str_val = {s = 0x0, len = -1074790400}, blob_val = {s = 0x0, len = -1074790400}, bitmap_val = 0}}, {type = DB1_STR, nul = 0, free = 140, val = {int_val = -1244124808,
              ll_val = 179144501624, double_val = 8.85091438937698e-313, time_val = -1244124808, string_val = 0xb5d82978 "ae372329-92b284e4-cb1aa8ab at 193.157.149.56", str_val = {s = 0xb5d82978 "ae372329-92b284e4-cb1aa8ab at 193.157.149.56", len = 41}, blob_val = {
                s = 0xb5d82978 "ae372329-92b284e4-cb1aa8ab at 193.157.149.56", len = 41}, bitmap_val = 3050842488}}, {type = DB1_INT, nul = 0, free = 16594016, val = {int_val = 2, ll_val = 2, double_val = 9.8813129168249309e-324, time_val = 2,
              string_val = 0x2 <Address 0x2 out of bounds>, str_val = {s = 0x2 <Address 0x2 out of bounds>, len = 0}, blob_val = {s = 0x2 <Address 0x2 out of bounds>, len = 0}, bitmap_val = 2}}, {type = DB1_INT, nul = 0, free = 15, val = {int_val = 0,
              ll_val = 30064771072, double_val = 1.4853970536756906e-313, time_val = 0, string_val = 0x0, str_val = {s = 0x0, len = 7}, blob_val = {s = 0x0, len = 7}, bitmap_val = 0}}, {type = DB1_INT, nul = 0, free = 234, val = {int_val = 0,
              ll_val = 30923764531200, double_val = 1.5278369694949961e-310, time_val = 0, string_val = 0x0, str_val = {s = 0x0, len = 7200}, blob_val = {s = 0x0, len = 7200}, bitmap_val = 0}}, {type = DB1_STR, nul = 0, free = 20, val = {int_val = -1244124752,
              ll_val = 183439468976, double_val = 9.0631139712402749e-313, time_val = -1244124752, string_val = 0xb5d829b0 "PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054", str_val = {s = 0xb5d829b0 "PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054", len = 42}, blob_val = {
                s = 0xb5d829b0 "PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054", len = 42}, bitmap_val = 3050842544}}, {type = DB1_STR, nul = 1, free = -1210855736, val = {int_val = 16593448, ll_val = -5351701324369153496, double_val = -7.199759953567273e-50,
              time_val = 16593448, string_val = 0xfd3228 "\030\360\001", str_val = {s = 0xfd3228 "\030\360\001", len = -1246040064}, blob_val = {s = 0xfd3228 "\030\360\001", len = -1246040064}, bitmap_val = 16593448}}, {type = DB1_STR, nul = 1, free = 135809173,
            val = {int_val = 1915400, ll_val = 8226580360673800, double_val = 4.0644707389612159e-308, time_val = 1915400, string_val = 0x1d3a08 "\b\213E\b\211D$\004\213\203\\\376\377\377\211\004$\350\262\361\373\377\203\304\024[]?\266", str_val = {
                s = 0x1d3a08 "\b\213E\b\211D$\004\213\203\\\376\377\377\211\004$\350\262\361\373\377\203\304\024[]?\266", len = 1915400}, blob_val = {s = 0x1d3a08 "\b\213E\b\211D$\004\213\203\\\376\377\377\211\004$\350\262\361\373\377\203\304\024[]?\266",
                len = 1915400}, bitmap_val = 1915400}}, {type = DB1_STR, nul = 0, free = 3, val = {int_val = -1210919000, ll_val = 106163263400, double_val = 5.2451621296335371e-313, time_val = -1210919000, string_val = 0xb7d2d7a8 "udp:129.240.253.133:5060",
              str_val = {s = 0xb7d2d7a8 "udp:129.240.253.133:5060", len = 24}, blob_val = {s = 0xb7d2d7a8 "udp:129.240.253.133:5060", len = 24}, bitmap_val = 3084048296}}, {type = DB1_BITMAP, nul = 0, free = 20, val = {int_val = 8159, ll_val = 70920170733838303,
              double_val = 6.3704038403022417e-304, time_val = 8159, string_val = 0x1fdf <Address 0x1fdf out of bounds>, str_val = {s = 0x1fdf <Address 0x1fdf out of bounds>, len = 16512389}, blob_val = {s = 0x1fdf <Address 0x1fdf out of bounds>, len = 16512389},
              bitmap_val = 8159}}, {type = DB1_DATETIME, nul = 0, free = -1, val = {int_val = 1345641222, ll_val = -4640175359789835514, double_val = -0.026157741446890113, time_val = 1345641222, string_val = 0x5034db06 <Address 0x5034db06 out of bounds>,
              str_val = {s = 0x5034db06 <Address 0x5034db06 out of bounds>, len = -1080375016}, blob_val = {s = 0x5034db06 <Address 0x5034db06 out of bounds>, len = -1080375016}, bitmap_val = 1345641222}}, {type = DB1_STR, nul = 0, free = 4094496, val = {
              int_val = -1244124696, ll_val = 88950188520, double_val = 4.3947232338834434e-313, time_val = -1244124696, string_val = 0xb5d829e8 "uloc-5034db04-67d5-1", str_val = {s = 0xb5d829e8 "uloc-5034db04-67d5-1", len = 20}, blob_val = {
                s = 0xb5d829e8 "uloc-5034db04-67d5-1", len = 20}, bitmap_val = 3050842600}}, {type = DB1_INT, nul = 1, free = -1244124968, val = {int_val = 26, ll_val = -5343475980977569766, double_val = -2.5829373687857378e-49, time_val = 26,
              string_val = 0x1a <Address 0x1a out of bounds>, str_val = {s = 0x1a <Address 0x1a out of bounds>, len = -1244124952}, blob_val = {s = 0x1a <Address 0x1a out of bounds>, len = -1244124952}, bitmap_val = 26}}, {type = DB1_INT, nul = 0,
            free = -1210856068, val = {int_val = 0, ll_val = 3530822193757814784, double_val = 1.1319814752747268e-72, time_val = 0, string_val = 0x0, str_val = {s = 0x0, len = 822083604}, blob_val = {s = 0x0, len = 822083604}, bitmap_val = 0}}, {type = DB1_STR,
            nul = 0, free = -1244124964, val = {int_val = 12962280, ll_val = 85912308200, double_val = 4.2446320036545221e-313, time_val = 12962280, string_val = 0xc5c9e8 "hometest.voip.uio.no", str_val = {s = 0xc5c9e8 "hometest.voip.uio.no", len = 20},
              blob_val = {s = 0xc5c9e8 "hometest.voip.uio.no", len = 20}, bitmap_val = 12962280}}}
        nr_cols = <value optimized out>
#2  0x00fcfc5d in insert_ucontact (_r=0xfd3a00, _contact=0xb7d3ce90, _ci=0xc5cc80, _c=0xbf9acb84) at urecord.c:478
No locals.
#3  0x00c55bb0 in add_contacts (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:493
        res = <value optimized out>
        r = <value optimized out>
        u = 0xb7d3cc9c
#4  save (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:864
        c = 0xb7d3ce88
        st = <value optimized out>
        mode = <value optimized out>
        aor = {s = 0xc5c9e0 "2549619 at hometest.voip.uio.no", len = 28}
        ret = <value optimized out>
        u = <value optimized out>
#5  0x00c4dd43 in w_save2 (_m=0xb7d05e40, _d=0xb5d63060 "00?", _cflags=0x0) at reg_mod.c:407
No locals.
#6  0x080586ad in do_action (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1157
        ret = -5
        v = <value optimized out>
        dst = {send_sock = 0x8, to = {s = {sa_family = 0, sa_data = "\000\000h?\277\025\225-\000x.\031\n"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 3214593640}, sin_zero = "\025\225-\000x.\031\n"}, sin6 = {sin6_family = 0, sin6_port = 0,
              sin6_flowinfo = 3214593640, sin6_addr = {in6_u = {u6_addr8 = "\025\225-\000x.\031\n\000\000\000\000\b\000\000", u6_addr16 = {38165, 45, 11896, 2585, 0, 0, 8, 0}, u6_addr32 = {2987285, 169422456, 0, 8}}}, sin6_scope_id = 174394728}}, id = 2987257,
          proto = 1 '\001', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}}
        tmp = <value optimized out>
        new_uri = <value optimized out>
        end = <value optimized out>
        crt = <value optimized out>
        cmd = <value optimized out>
        len = <value optimized out>
        user = <value optimized out>
        uri = {user = {s = 0x1 <Address 0x1 out of bounds>, len = 174394568}, passwd = {s = 0x0, len = 10813760}, host = {s = 0xf <Address 0xf out of bounds>, len = 10813760}, port = {s = 0x961976 "\201\303~\326\016", len = 10813760}, params = {s = 0x0,
            len = -1080373724}, sip_params = {s = 0x0, len = 24}, headers = {s = 0x2d91be "\211?\366u\326?\377\377", <incomplete sequence \307>, len = 64768}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 294972, transport = {
            s = 0x81a4 <Address 0x81a4 out of bounds>, len = 1}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 2225}, method = {s = 0x0, len = 4096}, lr = {s = 0x10 <Address 0x10 out of bounds>, len = 0}, r2 = {
            s = 0x5034db04 <Address 0x5034db04 out of bounds>, len = 0}, gr = {s = 0x4e7c286b <Address 0x4e7c286b out of bounds>, len = 0}, transport_val = {s = 0x4e7c286b <Address 0x4e7c286b out of bounds>, len = 0}, ttl_val = {
            s = 0x4803c <Address 0x4803c out of bounds>, len = 0}, user_param_val = {s = 0xa201f9 "/etc/localtime", len = -1080373756}, maddr_val = {s = 0x964927 "\205\300\211\307tEe\203=\f", len = -1080373724}, method_val = {
            s = 0x962479 "\213]\364\213u\370\213}\374\211\354]?\223\314\025", len = 50}, lr_val = {s = 0xa4eff4 "\234M\025", len = 15}, r2_val = {s = 0xa201f9 "/etc/localtime", len = -1080373724}, gr_val = {
            s = 0x968b26 "\211\302\213]\364\211?u\370\213}\374\211\354]?\220\220\220\220\220\220\220\220U\211\345\203\354\030\211]\364\213E\f\350\217q\372\377\201?d\016", len = 168882984}}
        next_hop = {user = {s = 0xbf9accf8 "", len = 2970786}, passwd = {s = 0xb7c074f4 "\004", len = -1080369992}, host = {s = 0xbf9accf8 "", len = 135239351}, port = {s = 0xbf9adcb8 "\003", len = -1212126312}, params = {s = 0xb7d05e40 "\001", len = 5289495},
          sip_params = {s = 0x0, len = 1}, headers = {s = 0x4 <Address 0x4 out of bounds>, len = 173661209}, port_no = 52776, proto = 49050, type = 7504536, flags = 3214593272, transport = {s = 0x0, len = -1080369992}, ttl = {s = 0x0, len = -1080373848},
          user_param = {s = 0x80604e1 "\213\225T\377\377\377\213\215X\377\377\377\200\346\376\211N\b\t?V\004\201\304\300", len = -1080374024}, maddr = {s = 0xb7c07eb0 "q\003", len = -1211081152}, method = {s = 0xbf9acf08 "@?\277\340\337\244",
            len = -1212123916}, lr = {s = 0xb7c074f0 "\001", len = 0}, r2 = {s = 0xbf9acd10 "$?\277\240?", len = 10809332}, gr = {s = 0xbf9acd80 "", len = 10201187}, transport_val = {s = 0xa4eff4 "\234M\025", len = -1080373724}, ttl_val = {
            s = 0x97d2a0 "\205\300\017\204\033\001", len = 3}, user_param_val = {s = 0xa201f9 "/etc/localtime", len = -1080373888}, maddr_val = {s = 0xbf9acea4 "@^?(?\277\\\376\005\bH?\277\300\005?@^?", len = 9836918}, method_val = {
            s = 0x2df578 "\210", <incomplete sequence \324>, len = 4}, lr_val = {s = 0x728195 "x}", len = 9836918}, r2_val = {s = 0x2d5bb0 "\203\370\377\211\306\017\204i\373\377\377\213M\b\203\307\001\213A\030\003E\354\211A\030\017\266", len = 174394568},
          gr_val = {s = 0xa4f7240 "\020", len = 1}}
        u = <value optimized out>
        port = <value optimized out>
        i = <value optimized out>
        flags = <value optimized out>
        avp = <value optimized out>
        st = {flags = 3214593656, id = 5725, name = {n = 10617337, s = {s = 0xa201f9 "/etc/localtime", len = 3232052}, re = 0xa201f9}, avp = 0xf}
        sct = <value optimized out>
        sjt = <value optimized out>
        rve = <value optimized out>
        mct = <value optimized out>
        rv = <value optimized out>
        rv1 = <value optimized out>
        c1 = {cache_type = 10617337, val_type = 9949531, c = {avp_val = {n = 10809332, s = {s = 0xa4eff4 "\234M\025", len = 1}, re = 0xa4eff4}, pval = {rs = {s = 0xa4eff4 "\234M\025", len = 1}, ri = 168882984, flags = -1080373616}},
          i2s = "\342\302\227\000\371\001\242\000\000\000\000\000\000\000\000\000\371\001\242\000\234", <incomplete sequence \316>}
        s = {s = 0xa4eff4 "\234M\025", len = 10613054}
        srevp = {0xa50140, 0x97c1fd}
        mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {
            type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE,
            u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0,
              string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}}
#7  0x0805fe5c in run_actions (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1644
        t = 0xb7c405c0
        ret = <value optimized out>
        mod = <value optimized out>
        ms = 0
#8  0x080604e1 in run_actions_safe (h=0xbf9adcb8, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1708
        ctx = {rec_lev = 1, run_flags = 0, last_retcode = 0, jmp_env = {{__jmpbuf = {0, -1080369992, 0, -1080373464, -1167974323, 234811079}, __mask_was_saved = 0, __saved_mask = {__val = {174331556, 10813760, 3214593944, 10813796, 10813796, 173661228, 10813808,
                  10617337, 1, 10813760, 174339728, 13656, 10624544, 0, 842274453, 10809332, 174331548, 10613054, 3214594260, 9969796, 174331548, 8180, 10613054, 3214594308, 10806304, 3214594120, 10814568, 8216, 10809332, 8016, 174331528, 3214594116}}}}}
        ret = 0
#9  0x080f96b7 in rval_get_int (h=0xbf9adcb8, msg=0xbf9ac954, i=0xbf9ad268, rv=0xb7c406cc, cache=0x0) at rvalue.c:920
        r_avp = <value optimized out>
        avp_val = {n = -1212330376, s = {s = 0xb7bd4e78 "", len = 0}, re = 0xb7bd4e78}
        pval = {rs = {s = 0xb7bd4e78 "", len = -1211081152}, ri = 26, flags = -1080372456}
        tmp = {
          s = 0x80586ad "\213}\b1?\300\211E\360\017\224\302\tW\004\211G\b\243\300V&\b\351~\374\377\377\213]\f\213}\020\213K<\213S\030\211L$\b\213s0\211<$\211t$\004\353?]\f\213E\020\213SH\213s\030\211T$\f\213{<\211|$\b\213K0\211\004$\211L$\004\377V\004?]\f\213CT\213s\030\211D$\020\213SH\213E\020\211T$\f\213{<\211|$\b\213K0\211\004$\211L$\004\377V\004\351q\377\377\377\213]\f\213{`\213s\030\211|$\024\213KT\211L$\020\213CH\213M\020\211D$\f\213S<\211T$\b\213{0\211\f$\211|$\004\377V\004\351:\377\377\377\213M"...,
          len = -1211081152}
        s = <value optimized out>
        r = <value optimized out>
        ret = <value optimized out>
        destroy_pval = <value optimized out>
#10 0x080fcf38 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad268, rve=0xb7c406c8) at rvalue.c:1914
        i1 = <value optimized out>
        i2 = <value optimized out>
        ret = <value optimized out>
        c1 = {cache_type = 21, val_type = RV_NONE, c = {avp_val = {n = 42, s = {s = 0x2a <Address 0x2a out of bounds>, len = 13}, re = 0x2a}, pval = {rs = {s = 0x2a <Address 0x2a out of bounds>, len = 13}, ri = 15, flags = 22}},
          i2s = "\a\000\000\000p\000\000\000\003\000\000\000\352\000\000\000\001\000\000\000 \034"}
        c2 = {cache_type = 21, val_type = 173661376, c = {avp_val = {n = -1080372852, s = {s = 0xbf9ad18c "\250?\277zx\234", len = 10253525}, re = 0xbf9ad18c}, pval = {rs = {s = 0xbf9ad18c "\250?\277zx\234", len = 10253525}, ri = 174331536, flags = 174331536}},
          i2s = "\254\000\000\000\000@\000\000 \344\244\000\070rO\nX?\277\030", <incomplete sequence \321>}
        rv1 = <value optimized out>
        rv2 = <value optimized out>
#11 0x080fd009 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad544, rve=0xb7c40a70) at rvalue.c:1922
        i1 = <value optimized out>
        i2 = <value optimized out>
        ret = <value optimized out>
        c1 = {cache_type = 134610524, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 2}, re = 0x0}, pval = {rs = {s = 0x0, len = 2}, ri = 131072, flags = 137121601}}, i2s = "\030\b(\b\000\000\000\000\371\001\242\000P?\277 a1\000\b"}
        c2 = {cache_type = 3211677, val_type = 172978744, c = {avp_val = {n = 1, s = {s = 0x1 <Address 0x1 out of bounds>, len = -1080372644}, re = 0x1}, pval = {rs = {s = 0x1 <Address 0x1 out of bounds>, len = -1080372644}, ri = -1212121552, flags = 1}},
          i2s = "@^?\377\377\377\377\300\035\301\267\000\000\000\000\000\000\304\023\003"}
        rv1 = <value optimized out>
        rv2 = <value optimized out>
#12 0x080585fe in do_action (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1121
        ret = -5
        v = <value optimized out>
        dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = "\000\000\000\000\000\000\000\000\000\000\377\377\377\377"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\377\377\377\377"}, sin6 = {sin6_family = 0,
              sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {in6_u = {u6_addr8 = "\000\000\000\000\377\377\377\377\365\311?\v\000\000", u6_addr16 = {0, 0, 65535, 65535, 51701, 47043, 11, 0}, u6_addr32 = {0, 4294967295, 3083061749, 11}}}, sin6_scope_id = 0}},
          id = 0, proto = -99 '\235', send_flags = {f = 58 ':', blst_imask = 1 '\001'}}
        tmp = <value optimized out>
        new_uri = <value optimized out>
        end = <value optimized out>
        crt = <value optimized out>
        cmd = <value optimized out>
        len = <value optimized out>
        user = <value optimized out>
        uri = {user = {s = 0x15 <Address 0x15 out of bounds>, len = -1246040064}, passwd = {s = 0x1d38d8 "\a", len = -1080372168}, host = {
            s = 0x8184895 "\213E\360\203\304\034[^_\311?U\b\215F\b\215Y\370\211>\001?E\360\211\037\213B\b\203\300\b\211B\b\211D$\004\307\004$\006", len = 1915096}, port = {s = 0x1d38d8 "\a", len = 16384}, params = {s = 0xa4e420 "", len = 5}, sip_params = {
            s = 0xb5baf010 "\270\061\273\265\370?", len = -1244125016}, headers = {s = 0x0, len = 36}, port_no = 273, proto = 0, type = 3214595224, flags = 135776296, transport = {s = 0xb5baf000 "", len = 36}, ttl = {
            s = 0xffffffff <Address 0xffffffff out of bounds>, len = -1212370000}, user_param = {s = 0xb7baf008 "", len = 1630016}, maddr = {s = 0xbf9ad488 "\v", len = 0}, method = {s = 0xb7c40224 "\004", len = -1080369992}, lr = {
            s = 0xbf9ad4c8 "\330?\277\070\317\017\b\270?\277@^?\244\330\232\277$\002?", len = 135239351}, r2 = {s = 0xbf9adcb8 "\003", len = -1211891792}, gr = {s = 0xb7d05e40 "\001", len = -1244125004}, transport_val = {s = 0xbf9adcb8 "\003",
            len = -1080371552}, ttl_val = {s = 0xb <Address 0xb out of bounds>, len = 137187132}, user_param_val = {s = 0x111 <Address 0x111 out of bounds>, len = -1211905547}, maddr_val = {
            s = 0xbf9ad4c8 "\330?\277\070\317\017\b\270?\277@^?\244\330\232\277$\002?", len = 135778466}, method_val = {s = 0x111 <Address 0x111 out of bounds>, len = -1211905547}, lr_val = {s = 0xb <Address 0xb out of bounds>, len = 0}, r2_val = {s = 0x0,
            len = 135}, gr_val = {s = 0xb7d3d0a0 "\002", len = -1080371036}}
        next_hop = {user = {s = 0x2 <Address 0x2 out of bounds>, len = 160}, passwd = {s = 0xa641688 "", len = 9836918}, host = {
            s = 0xa50170 "\020Sn\n\020\337Y\n\210\026d\n\230\335Y\nh\016e\n0rO\n\240\re\nP\333Y\n\340\343\026\nPj\027\n\030hO\n\030hO\n\230\001\245", len = 0}, port = {s = 0xb7c121d4 "\004", len = -1080369992}, params = {s = 0xbf9ad3b8 "", len = 135239351},
          sip_params = {s = 0xbf9adcb8 "\003", len = -1212080704}, headers = {s = 0xb7d05e40 "\001", len = 5289495}, port_no = 0, proto = 0, type = 10813760, flags = 174331688, transport = {s = 0x54c0 <Address 0x54c0 out of bounds>, len = 10624544}, ttl = {
            s = 0x1 <Address 0x1 out of bounds>, len = 8}, user_param = {s = 0x0, len = -1080369992}, maddr = {s = 0x0, len = -1080372120}, method = {s = 0x80604e1 "\213\225T\377\377\377\213\215X\377\377\377\200\346\376\211N\b\t?V\004\201\304\300",
            len = -1080372296}, lr = {s = 0xb7c3ffb0 "\177\004", len = -1211081152}, r2 = {s = 0xbf9ad5c8 "", len = -1212079660}, gr = {s = 0xb7c121d0 "\001", len = 0}, transport_val = {s = 0x0, len = 1}, ttl_val = {s = 0x0, len = -1080369992}, user_param_val = {
            s = 0x0, len = -1080372328}, maddr_val = {s = 0xba623ddd <Address 0xba623ddd out of bounds>, len = 234811079}, method_val = {s = 0x0, len = 173660872}, lr_val = {s = 0xbf9ad404 "\330\070\035", len = -1080372216}, r2_val = {
            s = 0xa60ef2 "\211F\f1\300?\264&", len = -1212485624}, gr_val = {s = 0x0, len = 10809332}}
        u = <value optimized out>
        port = <value optimized out>
        i = <value optimized out>
        flags = <value optimized out>
        avp = <value optimized out>
        st = {flags = 4932939, id = 54696, name = {n = -1211904900, s = {s = 0xb7c3cc7c "\001", len = 0}, re = 0xb7c3cc7c}, avp = 0xbf9ad5d8}
        sct = <value optimized out>
        sjt = <value optimized out>
        rve = 0xb7c40a70
        mct = <value optimized out>
        rv = <value optimized out>
        rv1 = <value optimized out>
        c1 = {cache_type = 3083076132, val_type = 3083076128, c = {avp_val = {n = -1080371752, s = {s = 0xbf9ad5d8 "\270\330\232\277W\206\005\b\270?\277P\025?@^? \002?\004\333\064P", len = 135253816}, re = 0xbf9ad5d8}, pval = {rs = {
                s = 0xbf9ad5d8 "\270\330\232\277W\206\005\b\270?\277P\025?@^? \002?\004\333\064P", len = 135253816}, ri = -1080369992, flags = -1211081152}}, i2s = "\244\330\232\277$\002?\000\000\000\000\260\263\274\267@^?\362", <incomplete sequence \344>}
        s = {s = 0xb7c3cac0 "", len = 254}
        srevp = {0x80aaeec, 0xb7c3cc7c}
        mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {
            type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE,
            u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0,
              string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}}
#13 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1644
        t = 0xb7c41550
        ret = <value optimized out>
        mod = <value optimized out>
        ms = 0
#14 0x08058657 in do_action (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1136
        ret = 1
        v = 1
        dst = {send_sock = 0x0, to = {s = {sa_family = 320, sa_data = "\245\000\070\334Y\n\017\000\000\000@\001\245"}, sin = {sin_family = 320, sin_port = 165, sin_addr = {s_addr = 173661240}, sin_zero = "\017\000\000\000@\001\245"}, sin6 = {sin6_family = 320,
              sin6_port = 165, sin6_flowinfo = 173661240, sin6_addr = {in6_u = {u6_addr8 = "\017\000\000\000@\001\245\000v\031\226\000!?", u6_addr16 = {15, 0, 320, 165, 6518, 150, 52769, 149}, u6_addr32 = {15, 10813760, 9836918, 9817633}}},
              sin6_scope_id = 10813760}}, id = 0, proto = 64 '@', send_flags = {f = 1 '\001', blst_imask = 165 '\245'}}
        tmp = <value optimized out>
        new_uri = <value optimized out>
        end = <value optimized out>
        crt = <value optimized out>
        cmd = <value optimized out>
        len = <value optimized out>
        user = <value optimized out>
        uri = {user = {s = 0x0, len = 9949531}, passwd = {s = 0x1 <Address 0x1 out of bounds>, len = 24}, host = {s = 0x0, len = -1080371236}, port = {s = 0xb7d3d098 "\030\002", len = 10617337}, params = {s = 0x0, len = -1210855256}, sip_params = {
            s = 0xb7baf008 "", len = -1080371256}, headers = {s = 0x80fbc28 "\213]\364\213u\370\213}\374\311?\035\370]&\b\205\333\017\205*\377\377\377\241\370]&\b\213\065$\330&\b@\205\366\243\370]&\b\017\205k\001", len = -1212485624}, port_no = 53408,
          proto = 47059, type = 10616677, flags = 3214597412, transport = {s = 0xa6416a4 " /nett/kamailio/current/sbin/kamailio[26581]: DEBUG: db_cluster [dbcl_api.c:334]: parallel operation - cluster [data2] (7/0)\n", len = 0}, ttl = {
            s = 0x9ba863 "\207?\302\201", <incomplete sequence \372>, len = 10813796}, user_param = {s = 0x4e18e0 "193.157.149.56", len = 14}, maddr = {
            s = 0xa50170 "\020Sn\n\020\337Y\n\210\026d\n\230\335Y\nh\016e\n0rO\n\240\re\nP\333Y\n\340\343\026\nPj\027\n\030hO\n\030hO\n\230\001\245", len = 0}, method = {s = 0xb7bd578c "\004", len = -1080369992}, lr = {
            s = 0xbf9ad828 "8\331\232\277\070\317\017\b\270?\277@^?\004?\277\214W\275\267", len = 135239351}, r2 = {s = 0xbf9adcb8 "\003", len = -1212328320}, gr = {s = 0xb7d05e40 "\001", len = -1212373388}, transport_val = {s = 0x0, len = 0}, ttl_val = {
            s = 0xbf9adca8 "@^? ", <incomplete sequence \344\244>, len = -1}, user_param_val = {s = 0xb7bca674 "\002", len = -1080370172}, maddr_val = {s = 0xb7d3d0a0 "\002", len = -1080371184}, method_val = {s = 0xbf9add80 "i\362=\377", len = -1080371072},
          lr_val = {s = 0x9ba863 "\207?\302\201", <incomplete sequence \372>, len = -1212366860}, r2_val = {s = 0xb7bcbbdc "\a", len = 9949856}, gr_val = {s = 0x0, len = -1080370172}}
        next_hop = {user = {s = 0x2 <Address 0x2 out of bounds>, len = 0}, passwd = {s = 0x4e18e0 "193.157.149.56", len = 14}, host = {s = 0xe <Address 0xe out of bounds>, len = 0}, port = {s = 0xb7d3d0a8 "V\322\323\267\016", len = 0}, params = {
            s = 0xbf9ad718 "", len = 135249768}, sip_params = {s = 0xb7bcbff4 "\002", len = -1211081152}, headers = {s = 0x2 <Address 0x2 out of bounds>, len = -1212366860}, port_no = 0, proto = 0, type = 10618599, flags = 10618598, transport = {s = 0x0,
            len = -1212366860}, ttl = {s = 0xbf9ad924 "\001", len = -1210855264}, user_param = {s = 0x0, len = -1080369992}, maddr = {s = 0x0, len = -1080371256}, method = {
            s = 0x80604e1 "\213\225T\377\377\377\213\215X\377\377\377\200\346\376\211N\b\t?V\004\201\304\300", len = -1080371432}, lr = {s = 0xb7bd5680 "\002\002", len = -1211081152}, r2 = {s = 0x0, len = -1080371268}, gr = {s = 0x2 <Address 0x2 out of bounds>,
            len = 0}, transport_val = {s = 0x0, len = -1}, ttl_val = {s = 0x0, len = -1080369992}, user_param_val = {s = 0x0, len = -1080371464}, maddr_val = {s = 0xba62387d <Address 0xba62387d out of bounds>, len = 234811079}, method_val = {s = 0x0, len = 536},
          lr_val = {s = 0xb7d3d0a0 "\002", len = -1080371320}, r2_val = {s = 0x8184b77 "\241\004 &\b\213\230\260", len = 1629480}, gr_val = {s = 0x18dd28 "D$\b\213\203\304\376\377\377\211D$\004\377\223\300\376\377\377\205\300\017\204\264\002", len = 2}}
        u = <value optimized out>
        port = <value optimized out>
        i = <value optimized out>
        flags = <value optimized out>
        avp = <value optimized out>
        st = {flags = 24, id = 21712, name = {n = 64768, s = {s = 0xfd00 <Address 0xfd00 out of bounds>, len = 0}, re = 0xfd00}, avp = 0x0}
        sct = <value optimized out>
        sjt = <value optimized out>
        rve = 0xb7c40220
        mct = <value optimized out>
        rv = <value optimized out>
        rv1 = <value optimized out>
        c1 = {cache_type = 3082639244, val_type = 3082639240, c = {avp_val = {n = -1080370888, s = {s = 0xbf9ad938 "\030?\277=\205\005\b\270?\277P\026?@^?\210W\275\267\244\201", len = 135253816}, re = 0xbf9ad938}, pval = {rs = {
                s = 0xbf9ad938 "\030?\277=\205\005\b\270?\277P\026?@^?\210W\275\267\244\201", len = 135253816}, ri = -1080369992, flags = -1211081152}}, i2s = "\004?\277\214W\275\267\000\000\000\000\314\330\232\277\001\000\000\000\b "}
        s = {s = 0x1 <Address 0x1 out of bounds>, len = 0}
        srevp = {0x4803c, 0x81a4}
        mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {
            type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE,
            u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0,
              string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}}
#15 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1644
        t = 0xb7c41650
        ret = <value optimized out>
        mod = <value optimized out>
        ms = 0
#16 0x0805853d in do_action (h=0xbf9adcb8, a=0xb7bd64d8, msg=0xb7d05e40) at action.c:761
        ret = -5
        v = <value optimized out>
        dst = {send_sock = 0x96429f, to = {s = {sa_family = 65535, sa_data = "\377\377\300o\201\000\254\257\004\bpv\201"}, sin = {sin_family = 65535, sin_port = 65535, sin_addr = {s_addr = 8482752}, sin_zero = "\254\257\004\bpv\201"}, sin6 = {sin6_family = 65535,
              sin6_port = 65535, sin6_flowinfo = 8482752, sin6_addr = {in6_u = {u6_addr8 = "\254\257\004\bpv\201\000\000?\277H\216\200", u6_addr16 = {44972, 2052, 30320, 129, 56320, 49050, 36424, 128}, u6_addr32 = {134524844, 8484464, 3214597120, 8425032}}},
              sin6_scope_id = 8484904}}, id = -1208280984, proto = 1 '\001', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}}
        tmp = <value optimized out>
        new_uri = <value optimized out>
        end = <value optimized out>
        crt = <value optimized out>
        cmd = <value optimized out>
        len = <value optimized out>
        user = <value optimized out>
        uri = {user = {s = 0x1 <Address 0x1 out of bounds>, len = 8482752}, passwd = {s = 0x817828 "\314w\201", len = -1080370264}, host = {s = 0xbf9adbc4 "", len = 8407671}, port = {s = 0x804afac "\243\002", len = -1080370264}, params = {
            s = 0x8177cc "0\030\373\267\005", len = 3}, sip_params = {s = 0xb7fb1868 "\313`\005\b\020ii\r", len = 1}, headers = {s = 0x0, len = 1}, port_no = 65535, proto = 65535, type = ERROR_URI_T, flags = 168882984, transport = {
            s = 0xbf9adb6c "\364", <incomplete sequence \357\244>, len = 9945826}, ttl = {s = 0xa201f9 "/etc/localtime", len = 0}, user_param = {s = 0x0, len = -1080370252}, maddr = {
            s = 0xa50170 "\020Sn\n\020\337Y\n\210\026d\n\230\335Y\nh\016e\n0rO\n\240\re\nP\333Y\n\340\343\026\nPj\027\n\030hO\n\030hO\n\230\001\245", len = 8200}, method = {s = 0x2 <Address 0x2 out of bounds>, len = 9836918}, lr = {
            s = 0xa1ff65 "corrupted double-linked list", len = 3520}, r2 = {s = 0xa6416a4 " /nett/kamailio/current/sbin/kamailio[26581]: DEBUG: db_cluster [dbcl_api.c:334]: parallel operation - cluster [data2] (7/0)\n", len = 10813760}, gr = {
            s = 0xbf9adbf0 "$\032\220", len = 8484464}, transport_val = {s = 0x8050d9f "_setjmp", len = 10813760}, ttl_val = {s = 0xa50170 "\020Sn\n\020\337Y\n\210\026d\n\230\335Y\nh\016e\n0rO\n\240\re\nP\333Y\n\340\343\026\nPj\027\n\030hO\n\030hO\n\230\001\245",
            len = 10617337}, user_param_val = {s = 0xa8428197 <Address 0xa8428197 out of bounds>, len = 0}, maddr_val = {s = 0x0, len = 13656}, method_val = {s = 0xa21e20 "free(): corrupted unsorted chunks", len = 0}, lr_val = {
            s = 0x32341695 <Address 0x32341695 out of bounds>, len = 10809332}, r2_val = {s = 0xa64169c "", len = 10613054}, gr_val = {s = 0xbf9adc94 "(q\274\267h\335\232\277d\004\006\b\270?\277(q\274\267@^? ", <incomplete sequence \344\244>, len = 9969796}}
        next_hop = {user = {s = 0xa201f9 "/etc/localtime", len = -1080370596}, passwd = {s = 0xbf9ada20 "", len = 0}, host = {s = 0xbf9ada40 "\f\025\373\267\300o\201", len = 10805216}, port = {s = 0x3 <Address 0x3 out of bounds>, len = 168883016}, params = {
            s = 0x8 <Address 0x8 out of bounds>, len = 10613062}, sip_params = {s = 0x0, len = 8406405}, headers = {s = 0x90bcd0 "GLIBC_2.0", len = 134570187}, port_no = 6248, proto = 47099, type = 4294901762, flags = 8428425, transport = {
            s = 0x8050b70 "libc.so.6", len = -1208281844}, ttl = {s = 0x816fc0 "\370\276\001", len = -1208281028}, user_param = {s = 0xbf9ada98 "", len = -1080370484}, maddr = {s = 0x80495d "\205\300t\275\351=\376\377\377\215v", len = 10809332}, method = {
            s = 0xbf9adb78 "\224?\277\204 \230", len = 9973573}, lr = {s = 0xa6416a2 "42 /nett/kamailio/current/sbin/kamailio[26581]: DEBUG: db_cluster [dbcl_api.c:334]: parallel operation - cluster [data2] (7/0)\n", len = -1208281784}, r2 = {
            s = 0x8fa1a4 "\240\060\020D", len = 1}, gr = {s = 0x0, len = -1080370536}, transport_val = {s = 0xbf9adbb4 "\377\377\377\377\300o\201", len = -1472036457}, ttl_val = {s = 0x5 <Address 0x5 out of bounds>, len = 9428004}, user_param_val = {
            s = 0x8fc8c8 "\226\201B\250\017\204\035\346|`\234|\374Qcq\\zy\360\003\346`\033|\231/\310Km\301\243\353\372\266=/\222\210\353", <incomplete sequence \352>, len = -1472036458}, maddr_val = {s = 0x10 <Address 0x10 out of bounds>, len = 0}, method_val = {
            s = 0x0, len = 1}, lr_val = {s = 0x3e0 <Address 0x3e0 out of bounds>, len = -1208280984}, r2_val = {s = 0xb7fb1548 "", len = 134548895}, gr_val = {s = 0x906794 "", len = 134524844}}
        u = <value optimized out>
        port = <value optimized out>
        i = 12
        flags = <value optimized out>
        avp = <value optimized out>
        st = {flags = 1, id = 0, name = {n = 134548895, s = {s = 0x8050d9f "_setjmp", len = 0}, re = 0x8050d9f}, avp = 0x8261a74}
        sct = <value optimized out>
        sjt = <value optimized out>
        rve = <value optimized out>
        mct = <value optimized out>
        rv = 0xb7bc7128
        rv1 = <value optimized out>
        c1 = {cache_type = 174331548, val_type = 8180, c = {avp_val = {n = 10613054, s = {s = 0xa1f13e "%H:%M:%S", len = -1080369980}, re = 0xa1f13e}, pval = {rs = {s = 0xa1f13e "%H:%M:%S", len = -1080369980}, ri = 10806304, flags = 174331528}},
          i2s = "p\004\245\000\030 \000\000\364\357\244\000x\037\000\000$\032\220\000H\025"}
        s = {s = 0xbf9adcb8 "\003", len = 0}
        srevp = {0x1, 0x901a24}
        mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {
            type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE,
            u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0,
              string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}}
#17 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7bc7128, msg=0xb7d05e40) at action.c:1644
        t = 0xb7bd64d8
        ret = <value optimized out>
        mod = <value optimized out>
        ms = 0
#18 0x08060464 in run_top_route (a=0xb7bc7128, msg=0xb7d05e40, c=0x0) at action.c:1729
        ctx = {rec_lev = 3, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {-1080369992, 0, -1212387032, -1080370024, -1167969571, 234811079}, __mask_was_saved = 0, __saved_mask = {__val = {234, 1, 7200, 168883016, 10254458, 135, 4294967295, 12443959,
                  3050814344, 135, 3214597480, 135828890, 3083886144, 2, 0, 0, 7, 4532726, 3214597432, 4482608, 4809436, 4403718, 3214597448, 4591547, 0, 3084056472, 3214597480, 135297783, 3083886144, 1073741825, 0, 3083886144}}}}}
        p = 0xbf9adcb8
        ret = <value optimized out>
        sfbk = 0
#19 0x080e11e4 in receive_msg (buf=0x82c4f20 "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619 at hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"...,
    len=809, rcv_info=0xbf9ade98) at receive.c:209
        msg = <value optimized out>
        ctx = {rec_lev = 173660872, run_flags = 0, last_retcode = 0, jmp_env = {{__jmpbuf = {10809332, 21, 173660872, 10809332, 169272288, 10813760}, __mask_was_saved = -1080369688, __saved_mask = {__val = {9839737, 16384, 10806304, 0, 10809332, 169306704,
                  10813760, 3214597640, 9839737, 9448916, 3086685512, 169306712, 4294967295, 8482752, 134526028, 8484464, 3214597680, 8425032, 8484904, 3086686312, 1, 1, 0, 134549081, 0, 136715172, 1, 9448916, 10, 3083908104, 3214597892, 3214597912}}}}}
        ret = <value optimized out>
        inb = {s = 0x82c4f20 "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619 at hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"..., len = 809}
#20 0x08177f7b in udp_rcv_loop () at udp_server.c:544
        len = 809
        tmp = <value optimized out>
        fromlen = 16
        ri = {src_ip = {af = 2, len = 4, u = {addrl = {949329345, 3214597816, 9595014, 1575301381}, addr32 = {949329345, 3214597816, 9595014, 1575301381}, addr16 = {40385, 14485, 57016, 49050, 26758, 146, 12549, 24037},
              addr = "\301\235\225\070\270?\277\206h\222\000\005\061\345]"}}, dst_ip = {af = 2, len = 4, u = {addrl = {2248011905, 0, 0, 0}, addr32 = {2248011905, 0, 0, 0}, addr16 = {61569, 34301, 0, 0, 0, 0, 0, 0},
              addr = "\201\360\375\205", '\000' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023\304\301\235\225\070\000\000\000\000\000\000\000"}, sin = {
              sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 949329345}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 949329345, sin6_addr = {in6_u = {u6_addr8 = '\000' <repeats 15 times>,
                  u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0xb7d2d8e8, proto = 1 '\001'}
        buf = "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619 at hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"...
#21 0x080b0426 in main_loop () at main.c:1633
        i = 2
        pid = 0
        si = 0xb7d2d8e8
        si_desc = "udp receiver child=2 sock=129.240.253.133:5060\000\277\225H\030\b\b\237\034\000\b\237\034\000\377\377\377\377\260\250 \b\001\000\000\000P\360\272\265`-?p\351\026\npI?\002\000\000\000p\351\026\n\004\333\064P\001\000\000\000\000\000\000\000\032\000\000\000\002", '\000' <repeats 11 times>"\350, \337\232\277"
        nrprocs = 6
#22 0x080b3d02 in main (argc=17, argv=0xbf9ae174) at main.c:2546
        cfg_stream = 0xa109008
        c = <value optimized out>
        r = 0
        tmp = 0xbf9ae1bc "\f\377\232\277.\377\232\277\071\377\232\277L\377\232\277o\377\232\277u\377\232\277\206\377\232\277\250\377\232\277\260\377\232\277"
        tmp_len = 9592645
        port = 136337865
        proto = -1080368920
        ret = <value optimized out>
        seed = 1746956911
        rfd = 4
        debug_save = 0
        debug_flag = 0
        dont_fork_cnt = 0
        n_lst = 0x0
        p = 0x0


> Also, can you give it a quick try with 9p instead of 7p for both 
> connections?

9p for both works! At least not crashes.

Changed to:

    modparam("db_cluster", "cluster", "data2=>voip2_data2=9s9p;voip1_data2=9s9p")
    modparam("usrloc", "db_mode",   3)

But it only writes to one of the databases. In this case the one defined in voip1_data2.
Have you tried db_cluster together with the registrar-module?
I'd like all writing to go to both databases, but reading should be failover-based.

-- 
?yvind Kolbu


From fborot at hotmail.com  Wed Aug 22 20:41:03 2012
From: fborot at hotmail.com (Fabian Borot)
Date: Wed, 22 Aug 2012 14:41:03 -0400
Subject: [SR-Users] SEAS Module
Message-ID: <SNT110-W575E7CED4DBC85AF16605ECDBF0@phx.gbl>



Hello, 

I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces  of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there.

I checked for information in previous versions only to find the same situation.(3.1 thru 3.3)

Is the information about the SEAS module specific to the WeSIP Application Server?
Thank you 

fborot 		 	   		  

From klaus.mailinglists at pernau.at  Thu Aug 23 07:54:11 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Thu, 23 Aug 2012 07:54:11 +0200
Subject: [SR-Users] Kamailio with asterisk for outbound calls
In-Reply-To: <5034D007.6040608@loopmethods.com>
References: <5034D007.6040608@loopmethods.com>
Message-ID: <5035C583.1090806@pernau.at>



On 22.08.2012 14:26, Vijay Thakur wrote:
> Hi All Kamailio Experts,
>
> I have configured Kamailio (kamailio 3.1.5) as media server.

Kamailio is a SIP proxy, not a media server. Maybe you mean that you are 
using Kamailio with rtpproxy as media relay.

> All things
> are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
> Calls. For this purpose i have followed the web page :

If you wan to you Asterisk as PSTN gateway only, then there is no need 
to follow this tutorial. This tutorial makes strong integration of 
Kamailio and Asterisk. For PSTN gateway functionality there is no need 
to integrate Kamailio and Asterisk - just configure Asterisk as gateway 
and forwards PSTN calls from Kamailio to Asterisk (and vice versa)

> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
> In this page, some points are not clear for me , as given below:
>
> (1) In case you use *sipregs* you have to create a record for each
> extension where to set the 'name' to value of 'name' from *sipusers*.
> The rest is populated by Asterisk from registrations.
 >
> (2) Be sure you configure Asterisk *to not authenticate* SIP requests
> coming from Kamailio.
>
> I am not sure that my local users chat is working through kamailio or
> asterisk, who is used for authorization.

What do you mean with "not sure"? For instant messaging between users 
there is no need to use Asterisk.

In above setup the authentication is done by Kamailio only.

regards
Klaus
> Any specific Web page to correct the issue will highly appreciated
> according to my scenario.
>
> Kindly guide me. Thanks in advance.
>
> --
> Best Regards,
>
> Vijay Thakur
> (Assistant Manager - Networks)
> Mobile   : +91 8744018065
> Mail     :vijay.thakur at loopmethods.com
>
> Loop IT Methods Private Limited
> 1st Floor, B-10, Sector-7, Noida, (U.P) India
> Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
> Fax: +91 971 728 330
> Web:www.loopmethods.com
>
> LOOP Disclaimer -------------------------------------------------------------------------------------------------
> This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited.
> -----------------------------------------------------------------------------------------------------------------
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From klaus.mailinglists at pernau.at  Thu Aug 23 07:57:54 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Thu, 23 Aug 2012 07:57:54 +0200
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
Message-ID: <5035C662.9000203@pernau.at>

The nonce is globally valid until it expires (for all kind of requests, 
IIRC you could also change the user).

Maybe your "internal" SIP clients calculates the response wrong. You 
could test the response calculation with this website (do not use 'real' 
passwords):
http://pernau.at/kd/sipdigest.php

klaus

On 22.08.2012 10:51, patrice.bodeven at orange.com wrote:
> Hello,
>
> I am working on Kamailio 3.2.2. There is no traffic, only functional
> test done.
>
> Based on the SIP Client used until now (Xlite), the INVITE is
> systematically authenticated by 407 as there is no Proxy-Authorization
> in the initial INVITE. Expected/normal behavior.
>
> But when using an internal SIP client, the initial INVITE is generated
> with a Proxy-Authorization containing the nonce value used by the
> successful registration.
>
> _In Authorization of Register (the answer is 200OK)_
>
> Digest username="+33296488922 at ims.v0.pftest.net",
> realm="sip:ims.v0.pftest.net",
> nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA",
> uri="sip:ims.v0.pftest.net",
> response="4ed51b8e62cf769b19f8c4771e068ad3",
> cnonce="06440000000529330fe1", qop=auth, nc=00000011
>
> _In Proxy-Authorization of Invite _
>
> Digest username="+33296488922 at ims.v0.pftest.net",
> realm="sip:ims.v0.pftest.net",
> nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA",
> uri="sip:0157361149 at ims.v0.pftest.net",
> response="b247052d6d5b37e0b6677c816390cb77",
> cnonce="06440000000529330fe1", qop=auth, nc=00000012
>
> As the nonce expire is high (see nonce_expire), I don't understand why
> Kamailio requests the Authentication of the INVITE by 407.
>
> _407 with Proxy-Authenticate_: Digest realm="ims.v0.pftest.net",
> nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true
>
> *Does it mean that the REGISTER and INVITE are using different Nonce
> value ? *
>
> ?Information on my configuration file.
>
> modparam("auth_db", "db_url", DBURLAUTH)
>
> modparam("auth", "auth_checks_register", 6) # callid and From TAG
>
> modparam("auth", "qop", "auth")    # enable qop=auth
>
> modparam("auth", "nonce_count", 1) # enable nonce_count support
>
> modparam("auth", "nonce_expire", 21600)   # Set nonce_expire to 6 hours
>
> For register, I do
>
>                   if (!www_authorize("$ru", "subscriber"))
>
>                  {
>
>                     if($? == -2 || $? == -3) {
>
>                        sl_send_reply("403","Forbidden auth ID");
>
>                       exit;
>
>                     }
>
>                     else {
>
>                          www_challenge("$ru", "17");
>
>                          exit;
>
>                     }
>
>                  }
>
> For invite, I do
>
>             if (!proxy_authorize("$fd", "subscriber"))
>
>             {
>
>                 if($? == -2 || $? == -3) {
>
>                    sl_send_reply("403","Forbidden auth ID");
>
>                    exit;
>
>                 }
>
>                else {
>
>                     proxy_challenge("$fd", "17");
>
>                     xlog("L_INFO","INVITE challenged by 407 to :$di:$dp
> \n");
>
>                     exit;
>
>                  }
>
>             }
>
> Followed by consume_credentials();
>
> Thanks in advance for your support.
>
> *Patrice B*
>
> _________________________________________________________________________________________________________________________
>
> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
> This message and its attachments may contain confidential or privileged information that may be protected by law;
> they should not be distributed, used or copied without authorisation.
> If you have received this email in error, please notify the sender and delete this message and its attachments.
> As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified.
> Thank you.
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From vijay.thakur at loopmethods.com  Thu Aug 23 08:31:40 2012
From: vijay.thakur at loopmethods.com (Vijay Thakur)
Date: Thu, 23 Aug 2012 12:01:40 +0530
Subject: [SR-Users] Kamailio with asterisk for outbound calls
In-Reply-To: <5035C583.1090806@pernau.at>
References: <5034D007.6040608@loopmethods.com> <5035C583.1090806@pernau.at>
Message-ID: <5035CE4C.7030702@loopmethods.com>

Thanks for clearing the doubts. You are very right, i am using kamailio 
as Media Relay.
Can you send me some specific document URL, from where i can configure 
Asterisk as PSTN Gateway.
Can we set Kamailio and Asterisk in one server.

Thanks in advance.

Vijay

  Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote:
>
>
> On 22.08.2012 14:26, Vijay Thakur wrote:
>> Hi All Kamailio Experts,
>>
>> I have configured Kamailio (kamailio 3.1.5) as media server.
>
> Kamailio is a SIP proxy, not a media server. Maybe you mean that you 
> are using Kamailio with rtpproxy as media relay.
>
>> All things
>> are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
>> Calls. For this purpose i have followed the web page :
>
> If you wan to you Asterisk as PSTN gateway only, then there is no need 
> to follow this tutorial. This tutorial makes strong integration of 
> Kamailio and Asterisk. For PSTN gateway functionality there is no need 
> to integrate Kamailio and Asterisk - just configure Asterisk as 
> gateway and forwards PSTN calls from Kamailio to Asterisk (and vice 
> versa)
>
>> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. 
>>
>> In this page, some points are not clear for me , as given below:
>>
>> (1) In case you use *sipregs* you have to create a record for each
>> extension where to set the 'name' to value of 'name' from *sipusers*.
>> The rest is populated by Asterisk from registrations.
> >
>> (2) Be sure you configure Asterisk *to not authenticate* SIP requests
>> coming from Kamailio.
>>
>> I am not sure that my local users chat is working through kamailio or
>> asterisk, who is used for authorization.
>
> What do you mean with "not sure"? For instant messaging between users 
> there is no need to use Asterisk.
>
> In above setup the authentication is done by Kamailio only.
>
> regards
> Klaus
>> Any specific Web page to correct the issue will highly appreciated
>> according to my scenario.
>>
>> Kindly guide me. Thanks in advance.
>>
>> -- 
>> Best Regards,
>>
>> Vijay Thakur
>> (Assistant Manager - Networks)
>> Mobile   : +91 8744018065
>> Mail     :vijay.thakur at loopmethods.com
>>
>> Loop IT Methods Private Limited
>> 1st Floor, B-10, Sector-7, Noida, (U.P) India
>> Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 
>> 178 (AUS)
>> Fax: +91 971 728 330
>> Web:www.loopmethods.com
>>
>> LOOP Disclaimer 
>> -------------------------------------------------------------------------------------------------
>> This message (including any attachments) contains confidential 
>> information intended for a specific individual and purpose, and is 
>> protected by law. If you are not the intended recipient, you should 
>> delete this message and are hereby notified that any disclosure, 
>> copying, or distribution of this message, or the taking of any action 
>> based on it, is strictly prohibited.
>> ----------------------------------------------------------------------------------------------------------------- 
>>
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>

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From miconda at gmail.com  Thu Aug 23 09:29:10 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 23 Aug 2012 09:29:10 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120822163434.GH16374@putsch.kolbu.ws>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
Message-ID: <5035DBC6.9010909@gmail.com>

Hello,

On 8/22/12 6:34 PM, ?yvind Kolbu wrote:
> On 2012-08-22 at 16:01, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> can you send the output of 'bt full' from gdb?
> #0  0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334

I will look over it when I get the first chance.

> [...]
>> Also, can you give it a quick try with 9p instead of 7p for both
>> connections?
> 9p for both works! At least not crashes.
>
> Changed to:
>
>      modparam("db_cluster", "cluster", "data2=>voip2_data2=9s9p;voip1_data2=9s9p")
>      modparam("usrloc", "db_mode",   3)
>
> But it only writes to one of the databases. In this case the one defined in voip1_data2.
> Have you tried db_cluster together with the registrar-module?
> I'd like all writing to go to both databases, but reading should be failover-based.
I looked I my config used for testing during development and I have an 
use case with 9p/9p for two connections for write, tested with acc module.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From klaus.mailinglists at pernau.at  Thu Aug 23 11:42:55 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Thu, 23 Aug 2012 11:42:55 +0200
Subject: [SR-Users] Kamailio with asterisk for outbound calls
In-Reply-To: <5035CE4C.7030702@loopmethods.com>
References: <5034D007.6040608@loopmethods.com> <5035C583.1090806@pernau.at>
	<5035CE4C.7030702@loopmethods.com>
Message-ID: <5035FB1F.4010905@pernau.at>



On 23.08.2012 08:31, Vijay Thakur wrote:
> Thanks for clearing the doubts. You are very right, i am using kamailio
> as Media Relay.
> Can you send me some specific document URL, from where i can configure
> Asterisk as PSTN Gateway.

There is no such document. But configuring a PSTN gateway is already in 
the default configuration file. Just search in the deafult configuration 
file for "WITH_PSTN".

> Can we set Kamailio and Asterisk in one server.

Yes, thats no problem. Either use 2 IP addresses on the same server, one 
for Kamailio and one for Asterisk, or use the same IP address and 
different ports.

regards
Klaus

PS: If you are building a public SIP service it is a good idea to not 
use the default port 5060 to get rid of SIP port scanners.

>
> Thanks in advance.
>
> Vijay
>
>   Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote:
>>
>>
>> On 22.08.2012 14:26, Vijay Thakur wrote:
>>> Hi All Kamailio Experts,
>>>
>>> I have configured Kamailio (kamailio 3.1.5) as media server.
>>
>> Kamailio is a SIP proxy, not a media server. Maybe you mean that you
>> are using Kamailio with rtpproxy as media relay.
>>
>>> All things
>>> are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
>>> Calls. For this purpose i have followed the web page :
>>
>> If you wan to you Asterisk as PSTN gateway only, then there is no need
>> to follow this tutorial. This tutorial makes strong integration of
>> Kamailio and Asterisk. For PSTN gateway functionality there is no need
>> to integrate Kamailio and Asterisk - just configure Asterisk as
>> gateway and forwards PSTN calls from Kamailio to Asterisk (and vice
>> versa)
>>
>>> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
>>>
>>> In this page, some points are not clear for me , as given below:
>>>
>>> (1) In case you use *sipregs* you have to create a record for each
>>> extension where to set the 'name' to value of 'name' from *sipusers*.
>>> The rest is populated by Asterisk from registrations.
>> >
>>> (2) Be sure you configure Asterisk *to not authenticate* SIP requests
>>> coming from Kamailio.
>>>
>>> I am not sure that my local users chat is working through kamailio or
>>> asterisk, who is used for authorization.
>>
>> What do you mean with "not sure"? For instant messaging between users
>> there is no need to use Asterisk.
>>
>> In above setup the authentication is done by Kamailio only.
>>
>> regards
>> Klaus
>>> Any specific Web page to correct the issue will highly appreciated
>>> according to my scenario.
>>>
>>> Kindly guide me. Thanks in advance.
>>>
>>> --
>>> Best Regards,
>>>
>>> Vijay Thakur
>>> (Assistant Manager - Networks)
>>> Mobile   : +91 8744018065
>>> Mail     :vijay.thakur at loopmethods.com
>>>
>>> Loop IT Methods Private Limited
>>> 1st Floor, B-10, Sector-7, Noida, (U.P) India
>>> Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011
>>> 178 (AUS)
>>> Fax: +91 971 728 330
>>> Web:www.loopmethods.com
>>>
>>> LOOP Disclaimer
>>> -------------------------------------------------------------------------------------------------
>>> This message (including any attachments) contains confidential
>>> information intended for a specific individual and purpose, and is
>>> protected by law. If you are not the intended recipient, you should
>>> delete this message and are hereby notified that any disclosure,
>>> copying, or distribution of this message, or the taking of any action
>>> based on it, is strictly prohibited.
>>> -----------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>


From martian at centrum.sk  Thu Aug 23 11:54:19 2012
From: martian at centrum.sk (martian at centrum.sk)
Date: Thu, 23 Aug 2012 11:54:19 +0200
Subject: [SR-Users] Possible bug in authentication
Message-ID: <20120823115419.63CC468C@centrum.sk>

Hello to everybody.
I am currently working with Kamailio 3.3.1 on RedHat.
The "loose_route" function was not working correctly and I observed some very strange behaviour (not as one described in the documentation of the function).
I have found that there needs to be a port included in the "alias" variable for the loose_route function to work correctly.
However, upon adding the port to alias, the INVITE messages were no longer authenticated (Kamailio just accepted them and didn't send proxy-auth header in 407 message).
?
My alias:
?
alias="domain.ch:5060"
?
Examining default routing logic, I found the problem here:
?
if (is_method("REGISTER") || from_uri==myself)
{
 # authenticate requests
 ...
}
?
The "from_uri==myself" was no longer evaluated as true, because there was a port at the end of the alias.
The FROM Header of the INVITE messages looks like:
?
From: "acc1" ;tag=12345
?
..so .. no port number there.
?
Btw, I have fixed this with replacing the "myself" list with my own defined variable MY_DOMAIN.
?
#!define MY_DOMAIN ".*@domain.ch"
?
So now the condition looks like this:
?
if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
{
...
}
?
?
I am not sure if this is a bug that needs to be fixed or not. I am just pointing my finger at it and I hope it will contribute to the development.
Also, a valid description of this behavior (when using port in alias) would be appreciated.
?
Cheers,
Martin

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From miconda at gmail.com  Thu Aug 23 12:21:03 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 23 Aug 2012 12:21:03 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <20120823115419.63CC468C@centrum.sk>
References: <20120823115419.63CC468C@centrum.sk>
Message-ID: <5036040F.3030908@gmail.com>

Hello,

On 8/23/12 11:54 AM, martian at centrum.sk wrote:
>
> Hello to everybody.
>
> I am currently working with Kamailio 3.3.1 on RedHat.
>
> The "loose_route" function was not working correctly and I observed 
> some very strange behaviour (not as one described in the documentation 
> of the function).
>
> I have found that there needs to be a port included in the "alias" 
> variable for the loose_route function to work correctly.
>
> However, upon adding the port to alias, the INVITE messages were no 
> longer authenticated (Kamailio just accepted them and didn't send 
> proxy-auth header in 407 message).
>
> My alias:
>
> alias="domain.ch:5060"
>
> Examining default routing logic, I found the problem here:
>
> if (is_method("REGISTER") || from_uri==myself)
>
> {
>
> # authenticate requests
>
> ...
>
> }
>
> The "from_uri==myself" was no longer evaluated as true, because there 
> was a port at the end of the alias.
>
> The FROM Header of the INVITE messages looks like:
>
> From: "acc1" <sip:acc1 at domain.ch>;tag=12345
>
> ..so .. no port number there.
>
> Btw, I have fixed this with replacing the "myself" list with my own 
> defined variable MY_DOMAIN.
>
> #!define MY_DOMAIN ".*@domain.ch"
>
> So now the condition looks like this:
>
> if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
>
> {
>
> ...
>
> }
>
> I am not sure if this is a bug that needs to be fixed or not. I am 
> just pointing my finger at it and I hope it will contribute to the 
> development.
>
> Also, a valid description of this behavior (when using port in alias) 
> would be appreciated.
>

if you enclose the value of the alias parameter in double quotes, then 
it is taken as string value. If you want to set it to a host:port, then 
remove the double quotes:

alias=domain.ch:5060


Why do you say the loose_route() was working strangely? Do you add the 
hostname as record-route, not the IP address? Detail more about what you 
think is wrong with record routing/loose routing.


Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From jiri at iptel.org  Thu Aug 23 13:32:25 2012
From: jiri at iptel.org (jiri at iptel.org)
Date: Thu, 23 Aug 2012 17:02:25 +0530
Subject: [SR-Users] =?koi8-r?b?8NLPxMHF1NPRIMvP1NTFxNYg1yA0MsvNIM/UIO3P?=
	=?koi8-r?b?08vX2Swg1SDHz9LOz8zZ1s7Px88gy9XSz9LUwQ==?=
Message-ID: <3764168567.916BR9U2874243@iejwlvfm.oomtizucqv.info>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  ????????? ????? www.????-???-?????.?? [...] 

Content analysis details:   (10.2 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 3.5 BAYES_99               BODY: Bayes spam probability is 99 to 100%
                            [score: 1.0000]
 0.8 RCVD_IN_SORBS_WEB      RBL: SORBS: sender is an abusable web server
                            [116.203.165.242 listed in dnsbl.sorbs.net]
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                           [116.203.165.242 listed in bb.barracudacentral.org]
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
                            [116.203.165.242 listed in zen.spamhaus.org]
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS
 0.0 UNPARSEABLE_RELAY      Informational: message has unparseable relay lines


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From: <jiri at iptel.org>,
	<jan at iptel.org>,
	<5060 at iptel.org>,
	<serusers at iptel.org>,
	<andrei at iptel.org>,
	<info at iptel.org>,
	<info at iptel.org>,
	<5060 at iptel.org>
Subject: =?koi8-r?B?8NLPxMHF1NPRIMvP1NTFxNYg1yA0MsvNIM/UIO3P08vX2Swg1SDHzw==?=	=?koi8-r?B?0s7PzNnWzs/HzyDL1dLP0tTB?=
Date: Thu, 23 Aug 2012 17:02:25 +0530
Size: 1352
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From martian at centrum.sk  Thu Aug 23 13:51:49 2012
From: martian at centrum.sk (martian at centrum.sk)
Date: Thu, 23 Aug 2012 13:51:49 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <5036040F.3030908@gmail.com>
References: <20120823115419.63CC468C@centrum.sk> <5036040F.3030908@gmail.com>
Message-ID: <20120823135149.427B12C2@centrum.sk>

Ok, so .. I have a session border controller device that is a contact point for my SIP domain (SRV record in DNS set to its IP). All the trafic goes through it and it does things like topology hiding etc.. The device forwards the INVITE messages to Kamailio, because of the routing.?
?
The loose_route was working strangely, because it did not behave as described in the documentation.
Here is the sip message that it was suppose to pass:
?
ACK sip:acc1 at domain.ch:5060 SIP/2.0
Via: SIP/2.0/UDP domain.ch;branch=z9hG4bKac386033013
Max-Forwards: 70
From: "acc2" ;tag=1c1749458918
To: ;tag=1c1892801634
Call-ID: 17494024742382012111116@
CSeq: 2 ACK
Contact: 
Route: 
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: SBC_DEVICE
Content-Length: 0
?
As you can see, there is a Route header and a To_tag .. so the loose_route function should return true. But instead, it returned false, then t_check_trans() also returned false and the routing logic exited (exit;).
?
This happens when the value of alias is not enclosed in double quotes.
?
PS.: There is a "-" symbol in the domain name. Can't that be a problem causing the need for the double quotes?
PS2: Should there be only a domain name in the alias? or also the hostname part? ... for example: ? domain.ch:5060 or server.domain.ch:5060
?
Martin
?
______________________________________________________________
> Od: "Daniel-Constantin Mierla" 
> Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" 
> D?tum: 23.08.2012 12:21
> Predmet: Re: [SR-Users] Possible bug in authentication
>
Hello,
 
On 8/23/12 11:54 AM, martian at centrum.sk wrote:Hello to everybody.
I am currently working with Kamailio 3.3.1 on RedHat.
The "loose_route" function was not working correctly and I observed some very strange behaviour (not as one described in the documentation of the function).
I have found that there needs to be a port included in the "alias" variable for the loose_route function to work correctly.
However, upon adding the port to alias, the INVITE messages were no longer authenticated (Kamailio just accepted them and didn't send proxy-auth header in 407 message).
?
My alias:
?
alias="domain.ch:5060"
?
Examining default routing logic, I found the problem here:
?
if (is_method("REGISTER") || from_uri==myself)
{
 # authenticate requests
 ...
}
?
The "from_uri==myself" was no longer evaluated as true, because there was a port at the end of the alias.
The FROM Header of the INVITE messages looks like:
?
From: "acc1" ;tag=12345
?
..so .. no port number there.
?
Btw, I have fixed this with replacing the "myself" list with my own defined variable MY_DOMAIN.
?
#!define MY_DOMAIN ".*@domain.ch"
?
So now the condition looks like this:
?
if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
{
...
}
?
?
I am not sure if this is a bug that needs to be fixed or not. I am just pointing my finger at it and I hope it will contribute to the development.
Also, a valid description of this behavior (when using port in alias) would be appreciated.

 if you enclose the value of the alias parameter in double quotes, then it is taken as string value. If you want to set it to a host:port, then remove the double quotes:
 
alias=domain.ch:5060

 Why do you say the loose_route() was working strangely? Do you add the hostname as record-route, not the IP address? Detail more about what you think is wrong with record routing/loose routing.
 

 Cheers,
 Daniel
 
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/micondaKamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
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From klaus.mailinglists at pernau.at  Thu Aug 23 15:04:41 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Thu, 23 Aug 2012 15:04:41 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <20120823135149.427B12C2@centrum.sk>
References: <20120823115419.63CC468C@centrum.sk> <5036040F.3030908@gmail.com>
	<20120823135149.427B12C2@centrum.sk>
Message-ID: <50362A69.8060807@pernau.at>

The Route URI (sent by SBC) must be identical to the Record-Route URI 
(inserted by Kamailio).

To find out why loose_route returns FALSE increase log-level. 
loose_route uses the "ismyself" function to evaluate if the Route header 
addresses this Kamailio server. And the "ismyself" is very verbose when 
doing this check.

regards
Klaus

On 23.08.2012 13:51, martian at centrum.sk wrote:
> Ok, so .. I have a session border controller device that is a contact
> point for my SIP domain (SRV record in DNS set to its IP). All the
> trafic goes through it and it does things like topology hiding etc.. The
> device forwards the INVITE messages to Kamailio, because of the routing.
>
> The loose_route was working strangely, because it did not behave as
> described in the documentation.
>
> Here is the sip message that it was suppose to pass:
>
> ACK sip:acc1 at domain.ch:5060 SIP/2.0
>
> Via: SIP/2.0/UDP domain.ch;branch=z9hG4bKac386033013
>
> Max-Forwards: 70
>
> From: "acc2" <sip:acc2 at domain.ch>;tag=1c1749458918
>
> To: <sip:acc1@<IP_ADRESS_OF_KAMAILIO>;user=phone>;tag=1c1892801634
>
> Call-ID: 17494024742382012111116@<IP_ADDRESS_OF_SBC>
>
> CSeq: 2 ACK
>
> Contact: <sip:acc2 at domain.ch:5060>
>
> Route: <sip:<IP_ADDRESS_OF_KAMAILIO>;lr=on>
>
> Supported: em,timer,replaces,path,resource-priority
>
> Allow:
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
>
> User-Agent: SBC_DEVICE
>
> Content-Length: 0
>
> As you can see, there is a Route header and a To_tag .. so the
> loose_route function should return true. But instead, it returned false,
> then t_check_trans() also returned false and the routing logic exited
> (exit;).
>
> This happens when the value of alias is not enclosed in double quotes.
>
> PS.: There is a "-" symbol in the domain name. Can't that be a problem
> causing the need for the double quotes?
>
> PS2: Should there be only a domain name in the alias? or also the
> hostname part? ... for example:   domain.ch:5060 or server.domain.ch:5060
>
> Martin
>
> ______________________________________________________________
>  > Od: "Daniel-Constantin Mierla" <miconda at gmail.com>
>  > Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List" <sr-users at lists.sip-router.org>
>  > D?tum: 23.08.2012 12:21
>  > Predmet: Re: [SR-Users] Possible bug in authentication
>  >
>
> Hello,
>
> On 8/23/12 11:54 AM, martian at centrum.sk <mailto:martian at centrum.sk> wrote:
>
>     Hello to everybody.
>
>     I am currently working with Kamailio 3.3.1 on RedHat.
>
>     The "loose_route" function was not working correctly and I observed
>     some very strange behaviour (not as one described in the
>     documentation of the function).
>
>     I have found that there needs to be a port included in the "alias"
>     variable for the loose_route function to work correctly.
>
>     However, upon adding the port to alias, the INVITE messages were no
>     longer authenticated (Kamailio just accepted them and didn't send
>     proxy-auth header in 407 message).
>
>     My alias:
>
>     alias="domain.ch:5060"
>
>     Examining default routing logic, I found the problem here:
>
>     if (is_method("REGISTER") || from_uri==myself)
>
>     {
>
>     # authenticate requests
>
>     ...
>
>     }
>
>     The "from_uri==myself" was no longer evaluated as true, because
>     there was a port at the end of the alias.
>
>     The FROM Header of the INVITE messages looks like:
>
>     From: "acc1" <sip:acc1 at domain.ch>;tag=12345
>
>     ..so .. no port number there.
>
>     Btw, I have fixed this with replacing the "myself" list with my own
>     defined variable MY_DOMAIN.
>
>     #!define MY_DOMAIN ".*@domain.ch" <mailto:.*@domain.ch>
>
>     So now the condition looks like this:
>
>     if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
>
>     {
>
>     ...
>
>     }
>
>     I am not sure if this is a bug that needs to be fixed or not. I am
>     just pointing my finger at it and I hope it will contribute to the
>     development.
>
>     Also, a valid description of this behavior (when using port in
>     alias) would be appreciated.
>
>
> if you enclose the value of the alias parameter in double quotes, then
> it is taken as string value. If you want to set it to a host:port, then
> remove the double quotes:
>
> alias=domain.ch:5060
>
>
> Why do you say the loose_route() was working strangely? Do you add the
> hostname as record-route, not the IP address? Detail more about what you
> think is wrong with record routing/loose routing.
>
>
> Cheers,
> Daniel
>
> -- Daniel-Constantin Mierla -http://www.asipto.comhttp://twitter.com/#!/miconda  -http://www.linkedin.com/in/micondaKamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>


From miconda at gmail.com  Fri Aug 24 09:22:13 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Aug 2012 09:22:13 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <5035DBC6.9010909@gmail.com>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com>
Message-ID: <50372BA5.5020003@gmail.com>

Hello,

can you print the content of cls in frame 0 inside gdb?

p *cls

I found and fixed a potential issue, by now in master:

http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=4d15ba97bab58108cfedc45158d90583f67cdadf

You can cherry pick it in branch 3.3 and try it, with:

git pull origin
git cherry-pick -x 4d15ba97bab58108cfedc45158d90583f67cdadf

But first get the content of cls, the fix I did was not convincing me it 
is the reason for what you reported.

Cheers,
Daniel

On 8/23/12 9:29 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 8/22/12 6:34 PM, ?yvind Kolbu wrote:
>> On 2012-08-22 at 16:01, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> can you send the output of 'bt full' from gdb?
>> #0  0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, 
>> _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334
>
> I will look over it when I get the first chance.
>
>> [...]
>>> Also, can you give it a quick try with 9p instead of 7p for both
>>> connections?
>> 9p for both works! At least not crashes.
>>
>> Changed to:
>>
>>      modparam("db_cluster", "cluster", 
>> "data2=>voip2_data2=9s9p;voip1_data2=9s9p")
>>      modparam("usrloc", "db_mode",   3)
>>
>> But it only writes to one of the databases. In this case the one 
>> defined in voip1_data2.
>> Have you tried db_cluster together with the registrar-module?
>> I'd like all writing to go to both databases, but reading should be 
>> failover-based.
> I looked I my config used for testing during development and I have an 
> use case with 9p/9p for two connections for write, tested with acc 
> module.
>
> Cheers,
> Daniel
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From oyvind.kolbu at usit.uio.no  Fri Aug 24 10:48:09 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Fri, 24 Aug 2012 10:48:09 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <50372BA5.5020003@gmail.com>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
Message-ID: <20120824084809.GI16374@putsch.kolbu.ws>

On 2012-08-24 at 09:22, Daniel-Constantin Mierla wrote:
> Hello,
> 
> can you print the content of cls in frame 0 inside gdb?
> 
> p *cls

(gdb) p *cls
$1 = {name = {s = 0xb7b5a958 "data2=>voip2_data2=8r9p", len = 5}, clsid = 1761460487, ref = 1, rlist = {{clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0,
      prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0},
      clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0,
        0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0,
        0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0xb7b5a7c8,
        0x0, 0x0, 0x0, 0x0}, clen = 1, prio = 8, mode = 114, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}}, wlist = {
    {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {
      clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0},
    {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {
      clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0},
    {clist = {0x0, 0x0, 0x0, 0x0, 0x0}, clen = 0, prio = 0, mode = 0, crt = 0}, {clist = {0xb7b5a7c8, 0x0, 0x0, 0x0, 0x0}, clen = 1, prio = 9, mode = 112,
      crt = 0}}, usedcon = 0xb7b5a7c8, next = 0x0}


> I found and fixed a potential issue, by now in master:
> 
> http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=4d15ba97bab58108cfedc45158d90583f67cdadf
> 
> You can cherry pick it in branch 3.3 and try it, with:
> 
> git pull origin
> git cherry-pick -x 4d15ba97bab58108cfedc45158d90583f67cdadf
> 
> But first get the content of cls, the fix I did was not convincing me it 
> is the reason for what you reported.

First impression is that everything is working now, thanks! At least
registrations are getting populated in both databases from both servers. As
I'm out traveling I havn't yet tested calling and whether or not the
failover works as expected. I assume setting usrloc's db_mode to 3 is
recommended in this kind of setup?

For now I'm using:
    modparam("db_cluster", "cluster","data2=>voip1_data2=9s9p;voip2_data2=8s9p")
on my voip1 machine, and the line below on the voip2 machine:
    modparam("db_cluster", "cluster","data2=>voip2_data2=9s9p;voip1_data2=8s9p")

The same configuration created coredumps on both machines previously.

-- 
?yvind Kolbu


From miconda at gmail.com  Fri Aug 24 12:48:09 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Aug 2012 12:48:09 +0200
Subject: [SR-Users] SEAS Module
In-Reply-To: <SNT110-W575E7CED4DBC85AF16605ECDBF0@phx.gbl>
References: <SNT110-W575E7CED4DBC85AF16605ECDBF0@phx.gbl>
Message-ID: <50375BE9.7020606@gmail.com>

Hello,

the images are located inside the source tree in 
modules_k/seas/doc/images/. Auto-generation of the html version of the 
readme for browsing online does not copy the images folder. I did it 
manually for kamailio 3.3.x and devel versions, you should see the 
images now.

Cheers,
Daniel

On 8/22/12 8:41 PM, Fabian Borot wrote:
>
> Hello,
>
> I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces  of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there.
>
> I checked for information in previous versions only to find the same situation.(3.1 thru 3.3)
>
> Is the information about the SEAS module specific to the WeSIP Application Server?
> Thank you
>
> fborot 		 	   		
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From martian at centrum.sk  Fri Aug 24 14:41:54 2012
From: martian at centrum.sk (martian at centrum.sk)
Date: Fri, 24 Aug 2012 14:41:54 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <50362A69.8060807@pernau.at>
References: <20120823115419.63CC468C@centrum.sk>, <5036040F.3030908@gmail.com>,
	<20120823135149.427B12C2@centrum.sk> <50362A69.8060807@pernau.at>
Message-ID: <20120824144154.3DAD39EA@centrum.sk>

The Route and Record-route headers are identical.


From agranig at sipwise.com  Fri Aug 24 18:53:34 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 24 Aug 2012 18:53:34 +0200
Subject: [SR-Users] Wrong index on expires column in pua module
Message-ID: <5037B18E.8090902@sipwise.com>

Hi guys,

In the pua table, there is a unique key on the "expires" column, which
seems wrong to me, because you'll sooner or later end up with different
pua entries having the same expiry time, no?

At least we ended up with messages like this in the logs, I think it's
caused by pua_reginfo tests:

ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry
'1345474412' for key 'expires_idx'
ERROR: <core> [db_query.c:210]: error while submitting query
ERROR: pua [pua.c:1171]: while inserting in db table pua

Any comments on that?

Andreas

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From fborot at hotmail.com  Fri Aug 24 18:57:08 2012
From: fborot at hotmail.com (Fabian Borot)
Date: Fri, 24 Aug 2012 12:57:08 -0400
Subject: [SR-Users] SEAS Module
In-Reply-To: <50375BE9.7020606@gmail.com>
References: <SNT110-W575E7CED4DBC85AF16605ECDBF0@phx.gbl>,
	<50375BE9.7020606@gmail.com>
Message-ID: <SNT110-W499BAC46B3ADACB96F1DF0CDBD0@phx.gbl>


thank you daniel,
I dont mean to sound like a pain in the neck but somehow the images are distorted, I used 3 browsers (IE, Fireofx, Chrome) and they look the same, streched, skewed, distorted
take a look please

----------------------------------------
> Date: Fri, 24 Aug 2012 12:48:09 +0200
> From: miconda at gmail.com
> To: sr-users at lists.sip-router.org
> CC: fborot at hotmail.com
> Subject: Re: [SR-Users] SEAS Module
>
> Hello,
>
> the images are located inside the source tree in
> modules_k/seas/doc/images/. Auto-generation of the html version of the
> readme for browsing online does not copy the images folder. I did it
> manually for kamailio 3.3.x and devel versions, you should see the
> images now.
>
> Cheers,
> Daniel
>
> On 8/22/12 8:41 PM, Fabian Borot wrote:
> >
> > Hello,
> >
> > I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there.
> >
> > I checked for information in previous versions only to find the same situation.(3.1 thru 3.3)
> >
> > Is the information about the SEAS module specific to the WeSIP Application Server?
> > Thank you
> >
> > fborot
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
> 		 	   		  

From bligthart at btlnet.co.uk  Fri Aug 24 20:07:54 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Fri, 24 Aug 2012 18:07:54 +0000
Subject: [SR-Users] Kamailio Crash using Cassandra 1.1.2 during
	SIP	unregister
In-Reply-To: <D8261FE064C78B44B77EF7C4569D6CBF0D11D150@DB3PRD0510MB380.eurprd05.prod.outlook.com>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11D150@DB3PRD0510MB380.eurprd05.prod.outlook.com>
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11DDC3@DB3PRD0510MB380.eurprd05.prod.outlook.com>

An update on this issue:

The schema that is used by the cassandra module seems to be missing a few fields, the correct schema for location should be:

*************************
callid(string) cflags(int) contact(string) cseq(int) expires(timestamp) flags(int) last_modified(int) methods(int) path(string) q(double) received(string) socket(string) user_agent(string) username(string) ruid(string) instance(string)
username
contact
*************************

The two fields : ruid and instance where missing, please append the documentation. (utils/kamctl/dbcassandra/openser/location)

Thanks,

Bou



From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Boudewyn Ligthart
Sent: 22 August 2012 16:34
To: sr-users at lists.sip-router.org
Subject: [SR-Users] Kamailio Crash using Cassandra 1.1.2 during SIP unregister


Hi,

Kamailio crash using Cassandra 1.1.2 during SIP unregister (registering works fine)

The version of Kamailio is the latest GIT version:

version: kamailio 3.4.0-dev3 (x86_64/linux) 10327c
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 10327c
compiled on 14:53:11 Aug 22 2012 with gcc 4.6.3

BT of Core :

Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `kamailio -ddddd'.
Program terminated with signal 11, Segmentation fault.
#0  fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527
527             if(unlikely(f->u.nxt_free!=NULL)) {
(gdb) bt
#0  fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527
#1  0x00007fa55d35ac53 in db_free_columns (_r=0x7fa55db38c30) at db_res.c:86
#2  0x00007fa55d83f87d in db_cassa_query (_h=<optimized out>, _k=0x7fa55adc6aac, _op=<optimized out>, _v=<optimized out>, _c=0x7fff72ef0e30, _n=<optimized out>,
    _nc=16, _o=0x7fa55afca990, _r=0x7fff72ef0f10) at dbcassa_base.cpp:659
#3  0x00007fa55adb58eb in db_load_urecord (_c=0x7fa55db218d0, _d=0x7fa5576eee38, _aor=0x7fff72ef11b0) at udomain.c:592
#4  0x00007fa55adb93cd in get_urecord (_d=<optimized out>, _aor=0x7fff72ef11b0, _r=0x7fff72ef1200) at udomain.c:1017
#5  0x00007fa55ab91698 in lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38, _uri=<optimized out>) at lookup.c:156
#6  0x00007fa55ab957f2 in w_lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38 "\350\355nW\245\177", _uri=<optimized out>) at reg_mod.c:437
#7  0x000000000041c624 in do_action (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1153
#8  0x000000000041b493 in run_actions (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1646
#9  0x0000000000423a24 in run_actions_safe (h=<optimized out>, a=<optimized out>, msg=<optimized out>) at action.c:1710
#10 0x00000000004d79a0 in rval_get_int (h=0x7fff72ef2790, msg=<optimized out>, i=0x7fff72ef1c30, rv=<optimized out>, cache=<optimized out>) at rvalue.c:920
#11 0x00000000004dc82c in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef1c30, rve=0x7fa55db23218) at rvalue.c:1914
#12 0x00000000004dc8f9 in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef20d0, rve=0x7fa55db238e8) at rvalue.c:1922
#13 0x000000000041cb66 in do_action (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1123
#14 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1646
#15 0x000000000041c7e1 in do_action (h=0x7fff72ef2790, a=0x7fa55db15b38, msg=0x7fa55db40c48) at action.c:763
#16 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db118c8, msg=0x7fa55db40c48) at action.c:1646
#17 0x0000000000423ab0 in run_top_route (a=0x7fa55db118c8, msg=0x7fa55db40c48, c=<optimized out>) at action.c:1731
#18 0x00000000004b39ee in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:209
#19 0x000000000054de1a in udp_rcv_loop () at udp_server.c:544
#20 0x0000000000478879 in main_loop () at main.c:1635
#21 0x000000000041b002 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2548

Thanks,

Bou
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From bligthart at btlnet.co.uk  Fri Aug 24 20:10:53 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Fri, 24 Aug 2012 18:10:53 +0000
Subject: [SR-Users] db_cassandra
In-Reply-To: <D8261FE064C78B44B77EF7C4569D6CBF0D11CD44@DB3PRD0510MB380.eurprd05.prod.outlook.com>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11ADB3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A46FE.7080403@1and1.ro>
	<D8261FE064C78B44B77EF7C4569D6CBF0D11AF60@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<502A5829.3090201@1and1.ro>
	<D8261FE064C78B44B77EF7C4569D6CBF0D11CD44@DB3PRD0510MB380.eurprd05.prod.outlook.com>
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11DDE2@DB3PRD0510MB380.eurprd05.prod.outlook.com>

Brief update:

Removed the domain parameter in the schema (subscriber). The registation now works.

Thanks,

Bou

From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Boudewyn Ligthart
Sent: 21 August 2012 16:45
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
Subject: Re: [SR-Users] db_cassandra

Hi,
  Kamailio now, connects to Cassandra ok. However the query against the column family
does not yield any results. Followed the docs to only create the column family inside the keyspace.


From miconda at gmail.com  Fri Aug 24 22:46:38 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Aug 2012 22:46:38 +0200
Subject: [SR-Users] Wrong index on expires column in pua module
In-Reply-To: <5037B18E.8090902@sipwise.com>
References: <5037B18E.8090902@sipwise.com>
Message-ID: <5037E82E.6050001@gmail.com>

Hello,

On 8/24/12 6:53 PM, Andreas Granig wrote:
> Hi guys,
>
> In the pua table, there is a unique key on the "expires" column, which
> seems wrong to me, because you'll sooner or later end up with different
> pua entries having the same expiry time, no?
>
> At least we ended up with messages like this in the logs, I think it's
> caused by pua_reginfo tests:
>
> ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry
> '1345474412' for key 'expires_idx'
> ERROR: <core> [db_query.c:210]: error while submitting query
> ERROR: pua [pua.c:1171]: while inserting in db table pua
>
> Any comments on that?
the unique constraint on expires is wrong, indeed. Perhaps a copy&paste 
mistake when adding the index on expires, which makes sense of db only 
mode to fetch/delete expired records. The index was added by Peter Dunkley.

You can go ahead and commit/backport.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Fri Aug 24 23:02:26 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Aug 2012 23:02:26 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <20120823135149.427B12C2@centrum.sk>
References: <20120823115419.63CC468C@centrum.sk> <5036040F.3030908@gmail.com>
	<20120823135149.427B12C2@centrum.sk>
Message-ID: <5037EBE2.4020601@gmail.com>

Hello,

On 8/23/12 1:51 PM, martian at centrum.sk wrote:
>
> Ok, so .. I have a session border controller device that is a contact 
> point for my SIP domain (SRV record in DNS set to its IP). All the 
> trafic goes through it and it does things like topology hiding etc.. 
> The device forwards the INVITE messages to Kamailio, because of the 
> routing.
>
> The loose_route was working strangely, because it did not behave as 
> described in the documentation.
>
> Here is the sip message that it was suppose to pass:
>
> ACK sip:acc1 at domain.ch:5060 SIP/2.0
>
> Via: SIP/2.0/UDP domain.ch;branch=z9hG4bKac386033013
>
> Max-Forwards: 70
>
> From: "acc2" <sip:acc2 at domain.ch>;tag=1c1749458918
>
> To: <sip:acc1@<IP_ADRESS_OF_KAMAILIO>;user=phone>;tag=1c1892801634
>
> Call-ID: 17494024742382012111116@<IP_ADDRESS_OF_SBC>
>
> CSeq: 2 ACK
>
> Contact: <sip:acc2 at domain.ch:5060>
>
> Route: <sip:<IP_ADDRESS_OF_KAMAILIO>;lr=on>
>
> Supported: em,timer,replaces,path,resource-priority
>
> Allow: 
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
>
> User-Agent: SBC_DEVICE
>
> Content-Length: 0
>
> As you can see, there is a Route header and a To_tag .. so the 
> loose_route function should return true. But instead, it returned 
> false, then t_check_trans() also returned false and the routing logic 
> exited (exit;).
>
if this is an ACK for a 200 ok and domain.ch is the domain of Kamailio, 
then the ACK is broken. The r-uri should be the contact address of 
callee (contact address from 200ok). The reason can be that the sbc 
changed the contact in 200 ok or the r-uri. Also, it could be that the 
UA is the broken device.

> This happens when the value of alias is not enclosed in double quotes.
>
> PS.: There is a "-" symbol in the domain name. Can't that be a problem 
> causing the need for the double quotes?
>

No, '-' in the hostname is valid. If it would not be valid, kamailio 
will not start, alias value is checked at startup.

> PS2: Should there be only a domain name in the alias? or also the 
> hostname part? ... for example:   domain.ch:5060 or server.domain.ch:5060
>

Alias value has pretty much same format as for listen, like 
'protocol:address:port'. protocol and port are option, address can be IP 
or hostname. So

alias=domain.ch:5060

is valid.

Cheers,
Daniel

> Martin
>
> ______________________________________________________________
> > Od: "Daniel-Constantin Mierla" <miconda at gmail.com>
> > Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) 
> - Users Mailing List" <sr-users at lists.sip-router.org>
> > D?tum: 23.08.2012 12:21
> > Predmet: Re: [SR-Users] Possible bug in authentication
> >
>
> Hello,
>
> On 8/23/12 11:54 AM, martian at centrum.sk <mailto:martian at centrum.sk> wrote:
>
>     Hello to everybody.
>
>     I am currently working with Kamailio 3.3.1 on RedHat.
>
>     The "loose_route" function was not working correctly and I
>     observed some very strange behaviour (not as one described in the
>     documentation of the function).
>
>     I have found that there needs to be a port included in the "alias"
>     variable for the loose_route function to work correctly.
>
>     However, upon adding the port to alias, the INVITE messages were
>     no longer authenticated (Kamailio just accepted them and didn't
>     send proxy-auth header in 407 message).
>
>     My alias:
>
>     alias="domain.ch:5060"
>
>     Examining default routing logic, I found the problem here:
>
>     if (is_method("REGISTER") || from_uri==myself)
>
>     {
>
>     # authenticate requests
>
>     ...
>
>     }
>
>     The "from_uri==myself" was no longer evaluated as true, because
>     there was a port at the end of the alias.
>
>     The FROM Header of the INVITE messages looks like:
>
>     From: "acc1" <sip:acc1 at domain.ch>;tag=12345
>
>     ..so .. no port number there.
>
>     Btw, I have fixed this with replacing the "myself" list with my
>     own defined variable MY_DOMAIN.
>
>     #!define MY_DOMAIN ".*@domain.ch" <mailto:.*@domain.ch>
>
>     So now the condition looks like this:
>
>     if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
>
>     {
>
>     ...
>
>     }
>
>     I am not sure if this is a bug that needs to be fixed or not. I am
>     just pointing my finger at it and I hope it will contribute to the
>     development.
>
>     Also, a valid description of this behavior (when using port in
>     alias) would be appreciated.
>
>
> if you enclose the value of the alias parameter in double quotes, then 
> it is taken as string value. If you want to set it to a host:port, 
> then remove the double quotes:
>
> alias=domain.ch:5060
>
>
> Why do you say the loose_route() was working strangely? Do you add the 
> hostname as record-route, not the IP address? Detail more about what 
> you think is wrong with record routing/loose routing.
>
>
> Cheers,
> Daniel
>
> -- Daniel-Constantin Mierla -http://www.asipto.comhttp://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/micondaKamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Fri Aug 24 23:10:41 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Aug 2012 23:10:41 +0200
Subject: [SR-Users] SEAS Module
In-Reply-To: <SNT110-W499BAC46B3ADACB96F1DF0CDBD0@phx.gbl>
References: <SNT110-W575E7CED4DBC85AF16605ECDBF0@phx.gbl>,
	<50375BE9.7020606@gmail.com>
	<SNT110-W499BAC46B3ADACB96F1DF0CDBD0@phx.gbl>
Message-ID: <5037EDD1.8050904@gmail.com>

Hello,

On 8/24/12 6:57 PM, Fabian Borot wrote:
> thank you daniel,
> I dont mean to sound like a pain in the neck but somehow the images are distorted, I used 3 browsers (IE, Fireofx, Chrome) and they look the same, streched, skewed, distorted
> take a look please
indeed, they look bad, even in the source tree viewed with an image 
viewer application.

I found them in good shape for openser 1.3.x and copied them from there 
in the online docs. Hopefully they were not changed since them. I will 
have the check the commit logs and import the good ones there.

Let me know if find some image missing or does not seem to be the right 
version.

Cheers,
Daniel

>
> ----------------------------------------
>> Date: Fri, 24 Aug 2012 12:48:09 +0200
>> From: miconda at gmail.com
>> To: sr-users at lists.sip-router.org
>> CC: fborot at hotmail.com
>> Subject: Re: [SR-Users] SEAS Module
>>
>> Hello,
>>
>> the images are located inside the source tree in
>> modules_k/seas/doc/images/. Auto-generation of the html version of the
>> readme for browsing online does not copy the images folder. I did it
>> manually for kamailio 3.3.x and devel versions, you should see the
>> images now.
>>
>> Cheers,
>> Daniel
>>
>> On 8/22/12 8:41 PM, Fabian Borot wrote:
>>> Hello,
>>>
>>> I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there.
>>>
>>> I checked for information in previous versions only to find the same situation.(3.1 thru 3.3)
>>>
>>> Is the information about the SEAS module specific to the WeSIP Application Server?
>>> Thank you
>>>
>>> fborot
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>> 		 	   		
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From fborot at hotmail.com  Fri Aug 24 23:35:50 2012
From: fborot at hotmail.com (Fabian Borot)
Date: Fri, 24 Aug 2012 17:35:50 -0400
Subject: [SR-Users] SEAS Module
In-Reply-To: <5037EDD1.8050904@gmail.com>
References: <SNT110-W575E7CED4DBC85AF16605ECDBF0@phx.gbl>,
	<50375BE9.7020606@gmail.com>
	<SNT110-W499BAC46B3ADACB96F1DF0CDBD0@phx.gbl>,
	<5037EDD1.8050904@gmail.com>
Message-ID: <SNT110-W39A09529E04CD77556CCBCCDBD0@phx.gbl>


thank you Daniel, they look fine now

----------------------------------------
> Date: Fri, 24 Aug 2012 23:10:41 +0200
> From: miconda at gmail.com
> To: sr-users at lists.sip-router.org
> CC: fborot at hotmail.com
> Subject: Re: [SR-Users] SEAS Module
>
> Hello,
>
> On 8/24/12 6:57 PM, Fabian Borot wrote:
> > thank you daniel,
> > I dont mean to sound like a pain in the neck but somehow the images are distorted, I used 3 browsers (IE, Fireofx, Chrome) and they look the same, streched, skewed, distorted
> > take a look please
> indeed, they look bad, even in the source tree viewed with an image
> viewer application.
>
> I found them in good shape for openser 1.3.x and copied them from there
> in the online docs. Hopefully they were not changed since them. I will
> have the check the commit logs and import the good ones there.
>
> Let me know if find some image missing or does not seem to be the right
> version.
>
> Cheers,
> Daniel
>
> >
> > ----------------------------------------
> >> Date: Fri, 24 Aug 2012 12:48:09 +0200
> >> From: miconda at gmail.com
> >> To: sr-users at lists.sip-router.org
> >> CC: fborot at hotmail.com
> >> Subject: Re: [SR-Users] SEAS Module
> >>
> >> Hello,
> >>
> >> the images are located inside the source tree in
> >> modules_k/seas/doc/images/. Auto-generation of the html version of the
> >> readme for browsing online does not copy the images folder. I did it
> >> manually for kamailio 3.3.x and devel versions, you should see the
> >> images now.
> >>
> >> Cheers,
> >> Daniel
> >>
> >> On 8/22/12 8:41 PM, Fabian Borot wrote:
> >>> Hello,
> >>>
> >>> I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there.
> >>>
> >>> I checked for information in previous versions only to find the same situation.(3.1 thru 3.3)
> >>>
> >>> Is the information about the SEAS module specific to the WeSIP Application Server?
> >>> Thank you
> >>>
> >>> fborot
> >>> _______________________________________________
> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> >>> sr-users at lists.sip-router.org
> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> --
> >> Daniel-Constantin Mierla - http://www.asipto.com
> >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
> >>
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
> 		 	   		  

From serusers at iptel.org  Mon Aug 27 03:35:29 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Sun, 26 Aug 2012 17:35:29 -0800
Subject: [SR-Users] Vacancy #371
Message-ID: <503ACEC8.708090@iptel.org>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  As a company we deal with advising other business and end
  users on reusable materials, green advancements and nontypical forms of energy
   and also produce and promote projects that are web media focused. [...] 

Content analysis details:   (10.3 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 3.6 HELO_DYNAMIC_IPADDR2   Relay HELO'd using suspicious hostname (IP addr
                            2)
 0.0 TVD_RCVD_IP            TVD_RCVD_IP
-0.0 SPF_PASS               SPF: sender matches SPF record
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [184.77.159.149 listed in bb.barracudacentral.org]
 1.7 URIBL_DBL_SPAM         Contains an URL listed in the DBL blocklist
                            [URIs: euroinjobs.com]
 1.7 URIBL_BLACK            Contains an URL listed in the URIBL blacklist
                            [URIs: euroinjobs.com]
 0.6 URIBL_PH_SURBL         Contains an URL listed in the PH SURBL blocklist
                            [URIs: euroinjobs.com]
-0.0 BAYES_40               BODY: Bayes spam probability is 20 to 40%
                            [score: 0.2745]
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
                            [184.77.159.149 listed in zen.spamhaus.org]
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS


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Subject: RE: Vacancy #371
Date: Sun, 26 Aug 2012 17:35:29 -0800
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From serusers at iptel.org  Mon Aug 27 05:22:55 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Mon, 27 Aug 2012 00:22:55 -0300
Subject: [SR-Users] Vacancy #120
Message-ID: <4E546D4A.103010@iptel.org>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  As a company we deal with advising other business and end
  users on reusable materials, green advancements and nontypical forms of energy
   and also produce and promote projects that are web media focused. [...] 

Content analysis details:   (15.1 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 3.6 HELO_DYNAMIC_IPADDR2   Relay HELO'd using suspicious hostname (IP addr
                            2)
 0.0 TVD_RCVD_IP            TVD_RCVD_IP
 0.7 SPF_SOFTFAIL           SPF: sender does not match SPF record (softfail)
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
                            [186.63.33.68 listed in zen.spamhaus.org]
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [186.63.33.68 listed in bb.barracudacentral.org]
 1.7 URIBL_DBL_SPAM         Contains an URL listed in the DBL blocklist
                            [URIs: euroinjobs.com]
 1.7 URIBL_BLACK            Contains an URL listed in the URIBL blacklist
                            [URIs: euroinjobs.com]
 0.6 URIBL_PH_SURBL         Contains an URL listed in the PH SURBL blocklist
                            [URIs: euroinjobs.com]
 0.8 BAYES_50               BODY: Bayes spam probability is 40 to 60%
                            [score: 0.4970]
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS


-------------- next part --------------
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From serusers at iptel.org  Mon Aug 27 07:43:27 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Mon, 27 Aug 2012 12:43:27 +0700
Subject: [SR-Users] Advisement and promotion company seeking out local
	affiliate
Message-ID: <4DD81092B95A72C1F7A4DE96E70FCE9B@nebpcadaotagndncpused.ameriton.com>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  As a company we deal with advising other business and end
  users on reusable materials, green advancements and nontypical forms of energy
   and also produce and promote projects that are web media focused. [...] 

Content analysis details:   (9.6 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 0.7 SPF_SOFTFAIL           SPF: sender does not match SPF record (softfail)
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
                            [139.195.43.246 listed in zen.spamhaus.org]
 1.7 URIBL_BLACK            Contains an URL listed in the URIBL blacklist
                            [URIs: euroinjobs.com]
 0.6 URIBL_PH_SURBL         Contains an URL listed in the PH SURBL blocklist
                            [URIs: euroinjobs.com]
 0.8 BAYES_50               BODY: Bayes spam probability is 40 to 60%
                            [score: 0.5000]
 1.7 URIBL_DBL_SPAM         Contains an URL listed in the DBL blocklist
                            [URIs: euroinjobs.com]
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS


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From miconda at gmail.com  Mon Aug 27 08:56:15 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 08:56:15 +0200
Subject: [SR-Users] spam messages breaking through filters
Message-ID: <503B1A0F.8050503@gmail.com>

Hello,

there were some spam messages going through our mailing list filters. I 
was updating the filters on mailman, but there are new and new ideas 
coming from this guys.

Sorry for the pollution and I wanted to say that these messages are 
marked as spam by spamassasin, you can filter them locally as well 
looking for the header

X-Spam-Level: ***

You can adjust the number of '*'.

Some email services do that automatically, like gmail.com for example.

Of course, if anyone has better ideas of protection, please write here.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Mon Aug 27 08:58:45 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 08:58:45 +0200
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
In-Reply-To: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
References: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
Message-ID: <503B1AA5.5090505@gmail.com>

Hello,

if you want to execute from a remote server, maybe it is better to use 
mi_datagram or xmlrpc modules to run MI/RPC commands.

Cheers,
Daniel


On 8/22/12 12:02 PM, phillman25 wrote:
> Dear List
>
> I am trying to run the below command from a REMOTE server where i use 
> Cacti to graph all my data. I am trying to graph on a per trunk basis.
>
> kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
>
> This command yields an output on the local server.
>
> Could someone perhaps point me in the right direction?
>
>
> Thanks!
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Mon Aug 27 09:04:49 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 09:04:49 +0200
Subject: [SR-Users] ADSL client-side problem
In-Reply-To: <4501cdc1-a894-4b5d-837c-7f68c136cbcd@ganesh-consulting.ch>
References: <4501cdc1-a894-4b5d-837c-7f68c136cbcd@ganesh-consulting.ch>
Message-ID: <503B1C11.1030409@gmail.com>

Hello,

what is the expire time of your registrations? 24 hours seems a lot to 
re-register.

 From kamailio point of view, the contact is expired based on 
un-registration or timeout. You can look with 'kamctl ul show' to see if 
the phone is still registered or not.

Anyhow, in such cases is good to set registration time quite small, say 
5 or 10 minutes, but if there is a problem in the client side, it is not 
much to do in server side.

What is the adsl routed, does it do SIP ALG? I have snom on ADSL using 
tcp, does not seem to make big problems.

Cheers,
Daniel

On 8/16/12 2:10 PM, Simon Hintermann wrote:
>
> Hello,
>
> I am experiencing some disconnections with my VOIP phones.
>
> Let me explain:
>
> My server is an Asterisk + Kamailio in a datacenter with a fixed IP, 
> and a bridge firewall in front of it, so no NAT here.
>
> My client (several offices) have usually no problem when they have a 
> fixed IP, but encounter fatal disconnection when they have a dynamic 
> IP. I had the case with Siemens, SNOM and Aastra phones, without any 
> difference. They are of course natted behind an ADSL router. I also 
> have clients doing load-balance between two ADSL lines, which is also 
> problematic.
>
> The only way for me to recover is to change the private IP of the VOIP 
> phones. Seems more like a TCP problem or a routing problem, but I am 
> no network guru.
>
> I have two ADSL lines to test it, and when I switch from one line to 
> another, every phone becomes "Not registered". Even falling back tot 
> he first line does not help recovering. I did not try to wait for 
> longer than 24 hours to recover.
>
> So my question is, more generally: is it possible to allow the clients 
> (VOIP phones) to re-connect with a new IP address without having 
> closed correctly the first connection (SIP session, I presume)? It 
> seems not by default, but I played with the timeouts server-side and 
> phone-side, without luck.
>
> I saw many posts with PBX having problems when behind a dynamic IP, 
> but nothing about the clients having dynamic addresses or 
> load-balancing. Normally, having a dynamic IP does not mean that your 
> IP will change ? on-the-fly ?, but each time you shut down and reboot 
> later your ADSL router, but we came to the conclusion that every 
> client NOT having a fixed IP was experiencing fatal disconnects. So I 
> managed to test and validate this behavior with our two ADSL lines.
>
> Hope someone can help me.
>
> Greetings
>
> Simon
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Mon Aug 27 09:09:25 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 09:09:25 +0200
Subject: [SR-Users] centos (fedora) service startup file
In-Reply-To: <502A1AF6.2000408@rodnoe.tv>
References: <501AA9D6.4040500@rodnoe.tv> <502A18B4.8040207@gmail.com>
	<502A1AF6.2000408@rodnoe.tv>
Message-ID: <503B1D25.4040709@gmail.com>

Hello,

On 8/14/12 11:31 AM, andre wrote:
> 14.08.2012 12:21, Daniel-Constantin Mierla ?????:
>> Hello,
>>
>> thanks, it was committed in git master branch, backports to follow 
>> before minor releases.
>>
>> Do you build your own rpms or you install from the opensuse build 
>> system?
> From the opensuse build system.

ok, I will update that system as well. is 'success' available in all 
distros using rpms (like opensuse)?

Cheers,
Daniel

>>
>> Cheers,
>> Daniel
>>
>> On 8/2/12 6:24 PM, andre wrote:
>>> Somwhere losted since openser 1.0 :)
>>>
>>>
>>> Please fix in:
>>>
>>> start() {
>>> echo -n $"Starting $prog: "
>>> .....
>>> [ $RETVAL = 0 ] && touch something
>>> to
>>> [ $RETVAL = 0 ] && touch someting && success
>>>
>>> I mean call function
>>> success
>>> in the end of the expression.
>>>
>>> Thanks in advance
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From serusers at iptel.org  Mon Aug 27 09:52:51 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Mon, 27 Aug 2012 09:52:51 +0200
Subject: [SR-Users] Advisement and promotion company seeking out local
	affiliate
Message-ID: <2069868956.Y2DQ3CTJ015399@izismdmuxsslvro.vdmzwysks.va>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  As a company we deal with advising other business and end
  users on reusable materials, green advancements and nontypical forms of energy
   and also produce and promote projects that are web media focused. [...] 

Content analysis details:   (7.0 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
 3.6 HELO_DYNAMIC_IPADDR2   Relay HELO'd using suspicious hostname (IP addr
                            2)
 0.0 TVD_RCVD_IP            TVD_RCVD_IP
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
                            [213.177.0.199 listed in zen.spamhaus.org]
-0.0 SPF_PASS               SPF: sender matches SPF record
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [213.177.0.199 listed in bb.barracudacentral.org]
 0.8 BAYES_50               BODY: Bayes spam probability is 40 to 60%
                            [score: 0.4633]
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS


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URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120827/a185d11f/attachment.eml>

From raman.kumar at tcs.com  Mon Aug 27 10:04:12 2012
From: raman.kumar at tcs.com (Raman Kumar)
Date: Mon, 27 Aug 2012 13:34:12 +0530
Subject: [SR-Users] SMS module & IMS
Message-ID: <OF3740938A.D4091C8A-ON65257A67.002B9C29-65257A67.002BDE1E@tcs.com>

Hi,

Can any one help me .. how to configure sms module?
what changes required in scscf.cfg or any other cfg also??


Thanks and regards,
Raman Kumar
=====-----=====-----=====
Notice: The information contained in this e-mail
message and/or attachments to it may contain 
confidential or privileged information. If you are 
not the intended recipient, any dissemination, use, 
review, distribution, printing or copying of the 
information contained in this e-mail message 
and/or attachments to it are strictly prohibited. If 
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From phillman25 at gmail.com  Mon Aug 27 11:21:37 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 27 Aug 2012 12:21:37 +0300
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
In-Reply-To: <503B1AA5.5090505@gmail.com>
References: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
	<503B1AA5.5090505@gmail.com>
Message-ID: <CAEprJzJzr-qGKh138V866=e3z5GAMMJBdJFipypWGsgiAe+10A@mail.gmail.com>

Thanks for the info Daniel.

I have implemented mi_datagram module and defined it with the following
parameters:

modparam("mi_datagram", "socket_name", "udp:2.2.2.2:8033")

When i run the below command from a remote server:

echo -ne ":address_dump:\n\n" | nc -w 1 -u 2.2.2.2 8033

i don't receive an output, but when i monitor all traffic on the local
machine:

ngrep -q -t -P "" -W byline -d any host 2.2.2.2 and udp

i observe the output:

U 2012/08/27 12:41:36.735094 1.1.1.1:55621 -> 2.2.2.2:8033
:address_dump:


U 2012/08/27 12:41:36.735201 2.2.2.2:47024 -> 1.1.1.1:55621
200 OK
  71 <1, xx.xx.xx.xx, 0> [1]
  72 <1, yy.yy.yy.yy, 0> [4]


Seems like the command is accepted from remote server and executed on local
machine but not relayed back to the remote server. Looks like mi_datagram
replies on different port other than 8033. Is there a way to force the
reply on source port 8033?

Thanks
Phillip






On Mon, Aug 27, 2012 at 9:58 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> if you want to execute from a remote server, maybe it is better to use
> mi_datagram or xmlrpc modules to run MI/RPC commands.
>
> Cheers,
> Daniel
>
>
>
> On 8/22/12 12:02 PM, phillman25 wrote:
>
> Dear List
>
>  I am trying to run the below command from a REMOTE server where i use
> Cacti to graph all my data. I am trying to graph on a per trunk basis.
>
>  kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
>
>  This command yields an output on the local server.
>
>  Could someone perhaps point me in the right direction?
>
>
>  Thanks!
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>
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From miconda at gmail.com  Mon Aug 27 12:08:46 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 12:08:46 +0200
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
In-Reply-To: <CAEprJzJzr-qGKh138V866=e3z5GAMMJBdJFipypWGsgiAe+10A@mail.gmail.com>
References: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
	<503B1AA5.5090505@gmail.com>
	<CAEprJzJzr-qGKh138V866=e3z5GAMMJBdJFipypWGsgiAe+10A@mail.gmail.com>
Message-ID: <503B472E.7010603@gmail.com>

Hello,

I am not sure it supports right now sending from the same socket, but it 
should send back to same ip and port from where it received the command. 
A patch it might be required to get it like that, so far I don't 
remember anyone willing to get such thing.

Btw, another alternative, you can eventually use sercmd with ctl module 
-- it works also over tcp (as well as udp). You can run any MI command 
via 'mi', like:

sercmd ... mi address_dump

Look at sercmd -h for the options.

So you don't need to build a script around nc.

Cheers,
Daniel

On 8/27/12 11:21 AM, phillman25 wrote:
> Thanks for the info Daniel.
>
> I have implemented mi_datagram module and defined it with the 
> following parameters:
>
> modparam("mi_datagram", "socket_name", "udp:2.2.2.2:8033 
> <http://2.2.2.2:8033>")
>
> When i run the below command from a remote server:
>
> echo -ne ":address_dump:\n\n" | nc -w 1 -u 2.2.2.2 8033
>
> i don't receive an output, but when i monitor all traffic on the local 
> machine:
>
> ngrep -q -t -P "" -W byline -d any host 2.2.2.2 and udp
>
> i observe the output:
>
> U 2012/08/27 12:41:36.735094 1.1.1.1:55621 <http://1.1.1.1:55621> -> 
> 2.2.2.2:8033 <http://2.2.2.2:8033>
> :address_dump:
>
>
> U 2012/08/27 12:41:36.735201 2.2.2.2:47024 <http://2.2.2.2:47024> -> 
> 1.1.1.1:55621 <http://1.1.1.1:55621>
> 200 OK
>   71 <1, xx.xx.xx.xx, 0> [1]
>   72 <1, yy.yy.yy.yy, 0> [4]
>
>
> Seems like the command is accepted from remote server and executed on 
> local machine but not relayed back to the remote server. Looks like 
> mi_datagram replies on different port other than 8033. Is there a way 
> to force the reply on source port 8033?
>
> Thanks
> Phillip
>
>
>
>
>
>
> On Mon, Aug 27, 2012 at 9:58 AM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     if you want to execute from a remote server, maybe it is better to
>     use mi_datagram or xmlrpc modules to run MI/RPC commands.
>
>     Cheers,
>     Daniel
>
>
>
>     On 8/22/12 12:02 PM, phillman25 wrote:
>>     Dear List
>>
>>     I am trying to run the below command from a REMOTE server where i
>>     use Cacti to graph all my data. I am trying to graph on a per
>>     trunk basis.
>>
>>     kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
>>
>>     This command yields an output on the local server.
>>
>>     Could someone perhaps point me in the right direction?
>>
>>
>>     Thanks!
>>
>>
>>     _______________________________________________
>>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>     sr-users at lists.sip-router.org  <mailto:sr-users at lists.sip-router.org>
>>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Mon Aug 27 15:25:43 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 15:25:43 +0200
Subject: [SR-Users] centos (fedora) service startup file
In-Reply-To: <503B252E.7020107@rodnoe.tv>
References: <501AA9D6.4040500@rodnoe.tv> <502A18B4.8040207@gmail.com>
	<502A1AF6.2000408@rodnoe.tv> <503B1D25.4040709@gmail.com>
	<503B252E.7020107@rodnoe.tv>
Message-ID: <503B7557.3010805@gmail.com>

Do not send private messages from the mailing list discussions.

Maybe others can jump in the thread and provide good information, if 
people from discussion cannot sort out things. Also, mailing list is 
archived online, so people can search later on topics they are 
interested in.

Cheers,
Daniel

On 8/27/12 9:43 AM, andre wrote:
> 27.08.2012 10:09, Daniel-Constantin Mierla ?????:
>> Hello,
>>
>> On 8/14/12 11:31 AM, andre wrote:
>>> 14.08.2012 12:21, Daniel-Constantin Mierla ?????:
>>>> Hello,
>>>>
>>>> thanks, it was committed in git master branch, backports to follow 
>>>> before minor releases.
>>>>
>>>> Do you build your own rpms or you install from the opensuse build 
>>>> system?
>>> From the opensuse build system.
>>
>> ok, I will update that system as well. is 'success' available in all 
>> distros using rpms (like opensuse)?
> Not shure about Suse. I even have no someone to ask
>
> Better add to RH distros only.
>>
>> Cheers,
>> Daniel
>>
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/2/12 6:24 PM, andre wrote:
>>>>> Somwhere losted since openser 1.0 :)
>>>>>
>>>>>
>>>>> Please fix in:
>>>>>
>>>>> start() {
>>>>> echo -n $"Starting $prog: "
>>>>> .....
>>>>> [ $RETVAL = 0 ] && touch something
>>>>> to
>>>>> [ $RETVAL = 0 ] && touch someting && success
>>>>>
>>>>> I mean call function
>>>>> success
>>>>> in the end of the expression.
>>>>>
>>>>> Thanks in advance
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>>>> list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Mon Aug 27 15:28:59 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 15:28:59 +0200
Subject: [SR-Users] SMS module & IMS
In-Reply-To: <OF3740938A.D4091C8A-ON65257A67.002B9C29-65257A67.002BDE1E@tcs.com>
References: <OF3740938A.D4091C8A-ON65257A67.002B9C29-65257A67.002BDE1E@tcs.com>
Message-ID: <503B761B.4020303@gmail.com>

Hello,

it is very unlikely that someone has a config file like you need, 
especially when you play with IMS extensions.

The better way to get help is to try and present here were you got 
stuck. Then people can give you hints to try further.

Cheers,
Daniel

On 8/27/12 10:04 AM, Raman Kumar wrote:
> Hi,
>
> Can any one help me .. how to configure sms module?
> what changes required in scscf.cfg or any other cfg also??
>
>
> Thanks and regards,
> Raman Kumar
>
> =====-----=====-----=====
> Notice: The information contained in this e-mail
> message and/or attachments to it may contain
> confidential or privileged information. If you are
> not the intended recipient, any dissemination, use,
> review, distribution, printing or copying of the
> information contained in this e-mail message
> and/or attachments to it are strictly prohibited. If
> you have received this communication in error,
> please notify us by reply e-mail or telephone and
> immediately and permanently delete the message
> and any attachments. Thank you
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From phillman25 at gmail.com  Mon Aug 27 16:06:43 2012
From: phillman25 at gmail.com (phillman25)
Date: Mon, 27 Aug 2012 17:06:43 +0300
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
In-Reply-To: <503B472E.7010603@gmail.com>
References: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
	<503B1AA5.5090505@gmail.com>
	<CAEprJzJzr-qGKh138V866=e3z5GAMMJBdJFipypWGsgiAe+10A@mail.gmail.com>
	<503B472E.7010603@gmail.com>
Message-ID: <CAEprJzL6hM_L_Y3dBk3NpMzaOF-XWuE05b77rG5+M3Rkp0GsFg@mail.gmail.com>

Thanks Daniel i was able to implement what you advised below and am able to
poll specific data into CACTI successfully.

Thanks again!
Phillip


On Mon, Aug 27, 2012 at 1:08 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> I am not sure it supports right now sending from the same socket, but it
> should send back to same ip and port from where it received the command. A
> patch it might be required to get it like that, so far I don't remember
> anyone willing to get such thing.
>
> Btw, another alternative, you can eventually use sercmd with ctl module --
> it works also over tcp (as well as udp). You can run any MI command via
> 'mi', like:
>
> sercmd ... mi address_dump
>
> Look at sercmd -h for the options.
>
> So you don't need to build a script around nc.
>
> Cheers,
> Daniel
>
>
> On 8/27/12 11:21 AM, phillman25 wrote:
>
> Thanks for the info Daniel.
>
>  I have implemented mi_datagram module and defined it with the following
> parameters:
>
>  modparam("mi_datagram", "socket_name", "udp:2.2.2.2:8033")
>
>  When i run the below command from a remote server:
>
>  echo -ne ":address_dump:\n\n" | nc -w 1 -u 2.2.2.2 8033
>
>  i don't receive an output, but when i monitor all traffic on the local
> machine:
>
>  ngrep -q -t -P "" -W byline -d any host 2.2.2.2 and udp
>
>  i observe the output:
>
>  U 2012/08/27 12:41:36.735094 1.1.1.1:55621 -> 2.2.2.2:8033
> :address_dump:
>
>
>  U 2012/08/27 12:41:36.735201 2.2.2.2:47024 -> 1.1.1.1:55621
> 200 OK
>   71 <1, xx.xx.xx.xx, 0> [1]
>   72 <1, yy.yy.yy.yy, 0> [4]
>
>
>  Seems like the command is accepted from remote server and executed on
> local machine but not relayed back to the remote server. Looks like
> mi_datagram replies on different port other than 8033. Is there a way to
> force the reply on source port 8033?
>
>  Thanks
> Phillip
>
>
>
>
>
>
> On Mon, Aug 27, 2012 at 9:58 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> if you want to execute from a remote server, maybe it is better to use
>> mi_datagram or xmlrpc modules to run MI/RPC commands.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>> On 8/22/12 12:02 PM, phillman25 wrote:
>>
>>  Dear List
>>
>>  I am trying to run the below command from a REMOTE server where i use
>> Cacti to graph all my data. I am trying to graph on a per trunk basis.
>>
>>  kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
>>
>>  This command yields an output on the local server.
>>
>>  Could someone perhaps point me in the right direction?
>>
>>
>>  Thanks!
>>
>>
>>  _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>
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From lfurrea at simplecs.net  Mon Aug 27 18:54:38 2012
From: lfurrea at simplecs.net (Luis Fernando Urrea)
Date: Mon, 27 Aug 2012 10:54:38 -0600
Subject: [SR-Users] DROUTING nub question
Message-ID: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>

I haven't been able to find the answer in the documentation.

It seems DROUTING does not take regex in the dr_rules table to match
routable numbers, are there any kind of wilcards that could be used.

How does one specify different rules for 8 digit numbers and 10 digit
numbers.

TIA


-- 
O: 4000-1020
C: 6040-3624
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From miconda at gmail.com  Mon Aug 27 22:08:59 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 22:08:59 +0200
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
Message-ID: <503BD3DB.6010608@gmail.com>

Hello,

On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
> I haven't been able to find the answer in the documentation.
>
> It seems DROUTING does not take regex in the dr_rules table to match 
> routable numbers, are there any kind of wilcards that could be used.
>
> How does one specify different rules for 8 digit numbers and 10 digit 
> numbers.
>
>
drouting uses longest prefix matching to select the rule.

Do you have rules only on length or also on prefix? If you provide a bit 
more details about what routing rules you are looking for, we may be 
able to provide hints about how to implement in the config file.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Mon Aug 27 22:14:49 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 22:14:49 +0200
Subject: [SR-Users] ADSL client-side problem
In-Reply-To: <503B90A0.3000905@ganesh-consulting.ch>
References: <4501cdc1-a894-4b5d-837c-7f68c136cbcd@ganesh-consulting.ch>
	<503B1C11.1030409@gmail.com>
	<503B90A0.3000905@ganesh-consulting.ch>
Message-ID: <503BD539.3040308@gmail.com>

Hello,

please do not send private messages from mailing lists discussions, keep 
the mailing list cc-ed so other can participate or learn from archives.

On 8/27/12 5:22 PM, Simon Hintermann wrote:
> Hello,
>
> I will try with a smaller "Subscription Expiry (s):" in my SNOM.
>
> Just to be clear with the method of test:
>
> - firewall redirects SNOM to my first ISP
> - telephone works perfectly, registered in Kamailio...
> - firewall redirects SNOM to my second ISP
> - telephone is unregistered in Kamailio, impossible to "re-register" 
> the identity1, says "network error"
> - only way to recover is to change the SNOM's IP address

what do you mean by 'firewall redirects Snom to Nth ISP'? Is the IP of 
snom static or taken via dhcp from adsl router?

Cheers,
Daniel

>
> Thanks for your support, greetings
>
> Simon
>
> Le 27.08.2012 09:04, Daniel-Constantin Mierla a ?crit :
>> Hello,
>>
>> what is the expire time of your registrations? 24 hours seems a lot 
>> to re-register.
>>
>> From kamailio point of view, the contact is expired based on 
>> un-registration or timeout. You can look with 'kamctl ul show' to see 
>> if the phone is still registered or not.
>>
>> Anyhow, in such cases is good to set registration time quite small, 
>> say 5 or 10 minutes, but if there is a problem in the client side, it 
>> is not much to do in server side.
>>
>> What is the adsl routed, does it do SIP ALG? I have snom on ADSL 
>> using tcp, does not seem to make big problems.
>>
>> Cheers,
>> Daniel
>>
>> On 8/16/12 2:10 PM, Simon Hintermann wrote:
>>>
>>> Hello,
>>>
>>> I am experiencing some disconnections with my VOIP phones.
>>>
>>> Let me explain:
>>>
>>> My server is an Asterisk + Kamailio in a datacenter with a fixed IP, 
>>> and a bridge firewall in front of it, so no NAT here.
>>>
>>> My client (several offices) have usually no problem when they have a 
>>> fixed IP, but encounter fatal disconnection when they have a dynamic 
>>> IP. I had the case with Siemens, SNOM and Aastra phones, without any 
>>> difference. They are of course natted behind an ADSL router. I also 
>>> have clients doing load-balance between two ADSL lines, which is 
>>> also problematic.
>>>
>>> The only way for me to recover is to change the private IP of the 
>>> VOIP phones. Seems more like a TCP problem or a routing problem, but 
>>> I am no network guru.
>>>
>>> I have two ADSL lines to test it, and when I switch from one line to 
>>> another, every phone becomes "Not registered". Even falling back tot 
>>> he first line does not help recovering. I did not try to wait for 
>>> longer than 24 hours to recover.
>>>
>>> So my question is, more generally: is it possible to allow the 
>>> clients (VOIP phones) to re-connect with a new IP address without 
>>> having closed correctly the first connection (SIP session, I 
>>> presume)? It seems not by default, but I played with the timeouts 
>>> server-side and phone-side, without luck.
>>>
>>> I saw many posts with PBX having problems when behind a dynamic IP, 
>>> but nothing about the clients having dynamic addresses or 
>>> load-balancing. Normally, having a dynamic IP does not mean that 
>>> your IP will change ? on-the-fly ?, but each time you shut down and 
>>> reboot later your ADSL router, but we came to the conclusion that 
>>> every client NOT having a fixed IP was experiencing fatal 
>>> disconnects. So I managed to test and validate this behavior with 
>>> our two ADSL lines.
>>>
>>> Hope someone can help me.
>>>
>>> Greetings
>>>
>>> Simon
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>> -- 
>> Daniel-Constantin Mierla -http://www.asipto.com
>> http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Mon Aug 27 22:19:08 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 22:19:08 +0200
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
In-Reply-To: <CAEprJzL6hM_L_Y3dBk3NpMzaOF-XWuE05b77rG5+M3Rkp0GsFg@mail.gmail.com>
References: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
	<503B1AA5.5090505@gmail.com>
	<CAEprJzJzr-qGKh138V866=e3z5GAMMJBdJFipypWGsgiAe+10A@mail.gmail.com>
	<503B472E.7010603@gmail.com>
	<CAEprJzL6hM_L_Y3dBk3NpMzaOF-XWuE05b77rG5+M3Rkp0GsFg@mail.gmail.com>
Message-ID: <503BD63C.9010809@gmail.com>

Hello,

On 8/27/12 4:06 PM, phillman25 wrote:
> Thanks Daniel i was able to implement what you advised below and am 
> able to poll specific data into CACTI successfully.
great! Have you made it like a plugin? Maybe you can write few words 
about or make web page available for it, there is a similar plugin for 
munin, published by Inaki:

- http://dev.sipdoc.net/projects/sip-stuff/wiki/KamailioMuninPlugin

Cheers,
Daniel

>
> Thanks again!
> Phillip
>
> On Mon, Aug 27, 2012 at 1:08 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     I am not sure it supports right now sending from the same socket,
>     but it should send back to same ip and port from where it received
>     the command. A patch it might be required to get it like that, so
>     far I don't remember anyone willing to get such thing.
>
>     Btw, another alternative, you can eventually use sercmd with ctl
>     module -- it works also over tcp (as well as udp). You can run any
>     MI command via 'mi', like:
>
>     sercmd ... mi address_dump
>
>     Look at sercmd -h for the options.
>
>     So you don't need to build a script around nc.
>
>     Cheers,
>     Daniel
>
>
>     On 8/27/12 11:21 AM, phillman25 wrote:
>>     Thanks for the info Daniel.
>>
>>     I have implemented mi_datagram module and defined it with the
>>     following parameters:
>>
>>     modparam("mi_datagram", "socket_name", "udp:2.2.2.2:8033
>>     <http://2.2.2.2:8033>")
>>
>>     When i run the below command from a remote server:
>>
>>     echo -ne ":address_dump:\n\n" | nc -w 1 -u 2.2.2.2 8033
>>
>>     i don't receive an output, but when i monitor all traffic on the
>>     local machine:
>>
>>     ngrep -q -t -P "" -W byline -d any host 2.2.2.2 and udp
>>
>>     i observe the output:
>>
>>     U 2012/08/27 12:41:36.735094 1.1.1.1:55621 <http://1.1.1.1:55621>
>>     -> 2.2.2.2:8033 <http://2.2.2.2:8033>
>>     :address_dump:
>>
>>
>>     U 2012/08/27 12:41:36.735201 2.2.2.2:47024 <http://2.2.2.2:47024>
>>     -> 1.1.1.1:55621 <http://1.1.1.1:55621>
>>     200 OK
>>       71 <1, xx.xx.xx.xx, 0> [1]
>>       72 <1, yy.yy.yy.yy, 0> [4]
>>
>>
>>     Seems like the command is accepted from remote server and
>>     executed on local machine but not relayed back to the remote
>>     server. Looks like mi_datagram replies on different port other
>>     than 8033. Is there a way to force the reply on source port 8033?
>>
>>     Thanks
>>     Phillip
>>
>>
>>
>>
>>
>>
>>     On Mon, Aug 27, 2012 at 9:58 AM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         if you want to execute from a remote server, maybe it is
>>         better to use mi_datagram or xmlrpc modules to run MI/RPC
>>         commands.
>>
>>         Cheers,
>>         Daniel
>>
>>
>>
>>         On 8/22/12 12:02 PM, phillman25 wrote:
>>>         Dear List
>>>
>>>         I am trying to run the below command from a REMOTE server
>>>         where i use Cacti to graph all my data. I am trying to graph
>>>         on a per trunk basis.
>>>
>>>         kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
>>>
>>>         This command yields an output on the local server.
>>>
>>>         Could someone perhaps point me in the right direction?
>>>
>>>
>>>         Thanks!
>>>
>>>
>>>         _______________________________________________
>>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>         sr-users at lists.sip-router.org  <mailto:sr-users at lists.sip-router.org>
>>>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>         -- 
>>         Daniel-Constantin Mierla -http://www.asipto.com
>>         http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>>         Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From lfurrea at simplecs.net  Mon Aug 27 22:20:26 2012
From: lfurrea at simplecs.net (Luis Fernando Urrea)
Date: Mon, 27 Aug 2012 14:20:26 -0600
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <503BD3DB.6010608@gmail.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
	<503BD3DB.6010608@gmail.com>
Message-ID: <CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>

Yes thank you Daniel,

On my setup I terminate local calls using colocated TDM equipment, these
calls are 8 digit numbers that start with 2,4,6,7 and 8

Then I use a FreeSWITCH box as a gateway for US and international calls, so
I was hoping I could somehow route based on numbers of digits.

I hope that is clear enough.

Regards,

Luis

On Mon, Aug 27, 2012 at 2:08 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

> Hello,
>
>
> On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
>
>> I haven't been able to find the answer in the documentation.
>>
>> It seems DROUTING does not take regex in the dr_rules table to match
>> routable numbers, are there any kind of wilcards that could be used.
>>
>> How does one specify different rules for 8 digit numbers and 10 digit
>> numbers.
>>
>>
>>  drouting uses longest prefix matching to select the rule.
>
> Do you have rules only on length or also on prefix? If you provide a bit
> more details about what routing rules you are looking for, we may be able
> to provide hints about how to implement in the config file.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda>
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
> http://asipto.com/u/kat
>
>


-- 
O: 2248-2483
C: 6040-3624
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From miconda at gmail.com  Mon Aug 27 22:30:33 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Aug 2012 22:30:33 +0200
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
	<503BD3DB.6010608@gmail.com>
	<CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>
Message-ID: <503BD8E9.80600@gmail.com>

Hello,

one solution is doing following trick:

- get the length of the dialed number via transformation and prefix it 
to the number

$rU = $(rU{s.len}) + $rU;

- in drouting keep rules only for prefixes  2, 4, 6, 7 and 8
- for drouting gateways set strip to 1, so the first digit (the length) 
is removed before forwarding

Cheers,
Daniel

On 8/27/12 10:20 PM, Luis Fernando Urrea wrote:
> Yes thank you Daniel,
>
> On my setup I terminate local calls using colocated TDM equipment, 
> these calls are 8 digit numbers that start with 2,4,6,7 and 8
>
> Then I use a FreeSWITCH box as a gateway for US and international 
> calls, so I was hoping I could somehow route based on numbers of digits.
>
> I hope that is clear enough.
>
> Regards,
>
> Luis
>
> On Mon, Aug 27, 2012 at 2:08 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>
>     On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
>
>         I haven't been able to find the answer in the documentation.
>
>         It seems DROUTING does not take regex in the dr_rules table to
>         match routable numbers, are there any kind of wilcards that
>         could be used.
>
>         How does one specify different rules for 8 digit numbers and
>         10 digit numbers.
>
>
>     drouting uses longest prefix matching to select the rule.
>
>     Do you have rules only on length or also on prefix? If you provide
>     a bit more details about what routing rules you are looking for,
>     we may be able to provide hints about how to implement in the
>     config file.
>
>     Cheers,
>     Daniel
>
>     -- 
>     Daniel-Constantin Mierla - http://www.asipto.com
>     http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
>     http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>     http://asipto.com/u/kat
>
>
>
>
> -- 
> O: 2248-2483
> C: 6040-3624
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From lfurrea at simplecs.net  Tue Aug 28 02:22:33 2012
From: lfurrea at simplecs.net (Luis Fernando Urrea)
Date: Mon, 27 Aug 2012 18:22:33 -0600
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <503BD8E9.80600@gmail.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
	<503BD3DB.6010608@gmail.com>
	<CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>
	<503BD8E9.80600@gmail.com>
Message-ID: <CAAPaQ=hDxSVo9gJeRz_JSC2Jm9N=0+jQ-ZMfyXUaPo90MEGu_A@mail.gmail.com>

Thanks Daniel,

That got me started, however, it seems that the return  value
from $(rU{s.len}) is actually an integer and therefore could not be
concatenated to $rU string ?

4(18194) ERROR: pv [pv_core.c:1798]: str value required to set R-URI user
14(18194) ERROR: <core> [lvalue.c:354]: setting pvar failed
14(18194) ERROR: <core> [lvalue.c:407]: assignment failed at pos:
(906,8-906,25)
14(18194) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg]
l=907 a=26 n=xlog
14(18194) ERROR: <script>: rU is now 8287825573!!!!!!!!!!!!!!!!

On Mon, Aug 27, 2012 at 2:30 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> one solution is doing following trick:
>
> - get the length of the dialed number via transformation and prefix it to
> the number
>
> $rU = $(rU{s.len}) + $rU;
>
> - in drouting keep rules only for prefixes  2, 4, 6, 7 and 8
> - for drouting gateways set strip to 1, so the first digit (the length) is
> removed before forwarding
>
> Cheers,
> Daniel
>
>
> On 8/27/12 10:20 PM, Luis Fernando Urrea wrote:
>
> Yes thank you Daniel,
>
>  On my setup I terminate local calls using colocated TDM equipment, these
> calls are 8 digit numbers that start with 2,4,6,7 and 8
>
>  Then I use a FreeSWITCH box as a gateway for US and international calls,
> so I was hoping I could somehow route based on numbers of digits.
>
>  I hope that is clear enough.
>
>  Regards,
>
>  Luis
>
> On Mon, Aug 27, 2012 at 2:08 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>> Hello,
>>
>>
>> On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
>>
>>> I haven't been able to find the answer in the documentation.
>>>
>>> It seems DROUTING does not take regex in the dr_rules table to match
>>> routable numbers, are there any kind of wilcards that could be used.
>>>
>>> How does one specify different rules for 8 digit numbers and 10 digit
>>> numbers.
>>>
>>>
>>>  drouting uses longest prefix matching to select the rule.
>>
>> Do you have rules only on length or also on prefix? If you provide a bit
>> more details about what routing rules you are looking for, we may be able
>> to provide hints about how to implement in the config file.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>> http://asipto.com/u/kat
>>
>>
>
>
>  --
> O: 2248-2483
> C: 6040-3624
>
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>


-- 
O: 2248-2483
C: 6040-3624
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From lfurrea at simplecs.net  Tue Aug 28 02:34:30 2012
From: lfurrea at simplecs.net (Luis Fernando Urrea)
Date: Mon, 27 Aug 2012 18:34:30 -0600
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <CAAPaQ=hDxSVo9gJeRz_JSC2Jm9N=0+jQ-ZMfyXUaPo90MEGu_A@mail.gmail.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
	<503BD3DB.6010608@gmail.com>
	<CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>
	<503BD8E9.80600@gmail.com>
	<CAAPaQ=hDxSVo9gJeRz_JSC2Jm9N=0+jQ-ZMfyXUaPo90MEGu_A@mail.gmail.com>
Message-ID: <CAAPaQ=jo=Sfr0ZF+R-H-1RWcXQ6rwqvW+M=JNwbFUnjQE7hLyw@mail.gmail.com>

I see there is a transformation to return the string value of an integer
but I can't seem to find the opposite one.

{s.int}
Return integer value of a string-represented number

On Mon, Aug 27, 2012 at 6:22 PM, Luis Fernando Urrea
<lfurrea at simplecs.net>wrote:

> Thanks Daniel,
>
> That got me started, however, it seems that the return  value
> from $(rU{s.len}) is actually an integer and therefore could not be
> concatenated to $rU string ?
>
> 4(18194) ERROR: pv [pv_core.c:1798]: str value required to set R-URI user
> 14(18194) ERROR: <core> [lvalue.c:354]: setting pvar failed
> 14(18194) ERROR: <core> [lvalue.c:407]: assignment failed at pos:
> (906,8-906,25)
> 14(18194) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg]
> l=907 a=26 n=xlog
> 14(18194) ERROR: <script>: rU is now 8287825573!!!!!!!!!!!!!!!!
>
> On Mon, Aug 27, 2012 at 2:30 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> one solution is doing following trick:
>>
>> - get the length of the dialed number via transformation and prefix it to
>> the number
>>
>> $rU = $(rU{s.len}) + $rU;
>>
>> - in drouting keep rules only for prefixes  2, 4, 6, 7 and 8
>> - for drouting gateways set strip to 1, so the first digit (the length)
>> is removed before forwarding
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/27/12 10:20 PM, Luis Fernando Urrea wrote:
>>
>> Yes thank you Daniel,
>>
>>  On my setup I terminate local calls using colocated TDM equipment,
>> these calls are 8 digit numbers that start with 2,4,6,7 and 8
>>
>>  Then I use a FreeSWITCH box as a gateway for US and international
>> calls, so I was hoping I could somehow route based on numbers of digits.
>>
>>  I hope that is clear enough.
>>
>>  Regards,
>>
>>  Luis
>>
>> On Mon, Aug 27, 2012 at 2:08 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
>>>
>>>> I haven't been able to find the answer in the documentation.
>>>>
>>>> It seems DROUTING does not take regex in the dr_rules table to match
>>>> routable numbers, are there any kind of wilcards that could be used.
>>>>
>>>> How does one specify different rules for 8 digit numbers and 10 digit
>>>> numbers.
>>>>
>>>>
>>>>  drouting uses longest prefix matching to select the rule.
>>>
>>> Do you have rules only on length or also on prefix? If you provide a bit
>>> more details about what routing rules you are looking for, we may be able
>>> to provide hints about how to implement in the config file.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.com
>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>> http://asipto.com/u/kat
>>>
>>>
>>
>>
>>  --
>> O: 2248-2483
>> C: 6040-3624
>>
>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>
>>
>
>
> --
> O: 2248-2483
> C: 6040-3624
>
>
>


-- 
O: 2248-2483
C: 6040-3624
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From jgallart at systemonenoc.com  Tue Aug 28 07:25:49 2012
From: jgallart at systemonenoc.com (Javi Gallart)
Date: Tue, 28 Aug 2012 07:25:49 +0200
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <CAAPaQ=jo=Sfr0ZF+R-H-1RWcXQ6rwqvW+M=JNwbFUnjQE7hLyw@mail.gmail.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
	<503BD3DB.6010608@gmail.com>
	<CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>
	<503BD8E9.80600@gmail.com>
	<CAAPaQ=hDxSVo9gJeRz_JSC2Jm9N=0+jQ-ZMfyXUaPo90MEGu_A@mail.gmail.com>
	<CAAPaQ=jo=Sfr0ZF+R-H-1RWcXQ6rwqvW+M=JNwbFUnjQE7hLyw@mail.gmail.com>
Message-ID: <503C565D.2000206@systemonenoc.com>

Hi

maybe this would work. You have a mask like "111111111111" (you choose 
the max length and the content). From the mask you extract as many 
digits as the length of your number with {s.substr,offset,length}, being 
length $(rU{s.len}}. Then you prepend the result to you $rU.

Hope it helps


Javi

On 08/28/2012 02:34 AM, Luis Fernando Urrea wrote:
> I see there is a transformation to return the string value of an 
> integer but I can't seem to find the opposite one.
>
> {s.int <http://s.int>}
> Return integer value of a string-represented number
>
> On Mon, Aug 27, 2012 at 6:22 PM, Luis Fernando Urrea 
> <lfurrea at simplecs.net <mailto:lfurrea at simplecs.net>> wrote:
>
>     Thanks Daniel,
>
>     That got me started, however, it seems that the return  value
>     from $(rU{s.len}) is actually an integer and therefore could not
>     be concatenated to $rU string ?
>
>     4(18194) ERROR: pv [pv_core.c:1798]: str value required to set
>     R-URI user
>     14(18194) ERROR: <core> [lvalue.c:354]: setting pvar failed
>     14(18194) ERROR: <core> [lvalue.c:407]: assignment failed at pos:
>     (906,8-906,25)
>     14(18194) ERROR: *** cfgtrace:
>     c=[/usr/local/etc/kamailio/kamailio.cfg] l=907 a=26 n=xlog
>     14(18194) ERROR: <script>: rU is now 8287825573!!!!!!!!!!!!!!!!
>
>     On Mon, Aug 27, 2012 at 2:30 PM, Daniel-Constantin Mierla
>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>         Hello,
>
>         one solution is doing following trick:
>
>         - get the length of the dialed number via transformation and
>         prefix it to the number
>
>         $rU = $(rU{s.len}) + $rU;
>
>         - in drouting keep rules only for prefixes  2, 4, 6, 7 and 8
>         - for drouting gateways set strip to 1, so the first digit
>         (the length) is removed before forwarding
>
>         Cheers,
>         Daniel
>
>
>         On 8/27/12 10:20 PM, Luis Fernando Urrea wrote:
>>         Yes thank you Daniel,
>>
>>         On my setup I terminate local calls using colocated TDM
>>         equipment, these calls are 8 digit numbers that start with
>>         2,4,6,7 and 8
>>
>>         Then I use a FreeSWITCH box as a gateway for US and
>>         international calls, so I was hoping I could somehow route
>>         based on numbers of digits.
>>
>>         I hope that is clear enough.
>>
>>         Regards,
>>
>>         Luis
>>
>>         On Mon, Aug 27, 2012 at 2:08 PM, Daniel-Constantin Mierla
>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>             Hello,
>>
>>
>>             On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
>>
>>                 I haven't been able to find the answer in the
>>                 documentation.
>>
>>                 It seems DROUTING does not take regex in the dr_rules
>>                 table to match routable numbers, are there any kind
>>                 of wilcards that could be used.
>>
>>                 How does one specify different rules for 8 digit
>>                 numbers and 10 digit numbers.
>>
>>
>>             drouting uses longest prefix matching to select the rule.
>>
>>             Do you have rules only on length or also on prefix? If
>>             you provide a bit more details about what routing rules
>>             you are looking for, we may be able to provide hints
>>             about how to implement in the config file.
>>
>>             Cheers,
>>             Daniel
>>
>>             -- 
>>             Daniel-Constantin Mierla - http://www.asipto.com
>>             http://twitter.com/#!/miconda
>>             <http://twitter.com/#%21/miconda> -
>>             http://www.linkedin.com/in/miconda
>>             Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>             http://asipto.com/u/kat
>>
>>
>>
>>
>>         -- 
>>         O: 2248-2483
>>         C: 6040-3624
>>
>>
>
>         -- 
>         Daniel-Constantin Mierla -http://www.asipto.com
>         http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>         Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>
>
>
>
>     -- 
>     O: 2248-2483
>     C: 6040-3624
>
>
>
>
>
> -- 
> O: 2248-2483
> C: 6040-3624
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



From miconda at gmail.com  Tue Aug 28 08:08:10 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Aug 2012 08:08:10 +0200
Subject: [SR-Users] DROUTING nub question
In-Reply-To: <503C565D.2000206@systemonenoc.com>
References: <CAAPaQ=hCg7j48fSw8dOOi7Fi4Dz155_HUOxud-JD2ML15CTPpg@mail.gmail.com>
	<503BD3DB.6010608@gmail.com>
	<CAAPaQ=jsETBWz5U9ScP4o2TZWPqEb-nSbW2Lw-B86-XQNEEKig@mail.gmail.com>
	<503BD8E9.80600@gmail.com>
	<CAAPaQ=hDxSVo9gJeRz_JSC2Jm9N=0+jQ-ZMfyXUaPo90MEGu_A@mail.gmail.com>
	<CAAPaQ=jo=Sfr0ZF+R-H-1RWcXQ6rwqvW+M=JNwbFUnjQE7hLyw@mail.gmail.com>
	<503C565D.2000206@systemonenoc.com>
Message-ID: <503C604A.2080209@gmail.com>

Hello,

this could be also an workaround for adding some prefix based on length.

But also the length can be used, it should work like:

$rU = "" + $(rU{s.len}) + $rU;

(note the empty string) or

pv_printf("$rU", "$(rU{s.len})$rU");

Cheers,
Daniel


On 8/28/12 7:25 AM, Javi Gallart wrote:
> Hi
>
> maybe this would work. You have a mask like "111111111111" (you choose 
> the max length and the content). From the mask you extract as many 
> digits as the length of your number with {s.substr,offset,length}, 
> being length $(rU{s.len}}. Then you prepend the result to you $rU.
>
> Hope it helps
>
>
> Javi
>
> On 08/28/2012 02:34 AM, Luis Fernando Urrea wrote:
>> I see there is a transformation to return the string value of an 
>> integer but I can't seem to find the opposite one.
>>
>> {s.int <http://s.int>}
>> Return integer value of a string-represented number
>>
>> On Mon, Aug 27, 2012 at 6:22 PM, Luis Fernando Urrea 
>> <lfurrea at simplecs.net <mailto:lfurrea at simplecs.net>> wrote:
>>
>>     Thanks Daniel,
>>
>>     That got me started, however, it seems that the return value
>>     from $(rU{s.len}) is actually an integer and therefore could not
>>     be concatenated to $rU string ?
>>
>>     4(18194) ERROR: pv [pv_core.c:1798]: str value required to set
>>     R-URI user
>>     14(18194) ERROR: <core> [lvalue.c:354]: setting pvar failed
>>     14(18194) ERROR: <core> [lvalue.c:407]: assignment failed at pos:
>>     (906,8-906,25)
>>     14(18194) ERROR: *** cfgtrace:
>>     c=[/usr/local/etc/kamailio/kamailio.cfg] l=907 a=26 n=xlog
>>     14(18194) ERROR: <script>: rU is now 8287825573!!!!!!!!!!!!!!!!
>>
>>     On Mon, Aug 27, 2012 at 2:30 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         one solution is doing following trick:
>>
>>         - get the length of the dialed number via transformation and
>>         prefix it to the number
>>
>>         $rU = $(rU{s.len}) + $rU;
>>
>>         - in drouting keep rules only for prefixes  2, 4, 6, 7 and 8
>>         - for drouting gateways set strip to 1, so the first digit
>>         (the length) is removed before forwarding
>>
>>         Cheers,
>>         Daniel
>>
>>
>>         On 8/27/12 10:20 PM, Luis Fernando Urrea wrote:
>>>         Yes thank you Daniel,
>>>
>>>         On my setup I terminate local calls using colocated TDM
>>>         equipment, these calls are 8 digit numbers that start with
>>>         2,4,6,7 and 8
>>>
>>>         Then I use a FreeSWITCH box as a gateway for US and
>>>         international calls, so I was hoping I could somehow route
>>>         based on numbers of digits.
>>>
>>>         I hope that is clear enough.
>>>
>>>         Regards,
>>>
>>>         Luis
>>>
>>>         On Mon, Aug 27, 2012 at 2:08 PM, Daniel-Constantin Mierla
>>>         <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>>
>>>             Hello,
>>>
>>>
>>>             On 8/27/12 6:54 PM, Luis Fernando Urrea wrote:
>>>
>>>                 I haven't been able to find the answer in the
>>>                 documentation.
>>>
>>>                 It seems DROUTING does not take regex in the dr_rules
>>>                 table to match routable numbers, are there any kind
>>>                 of wilcards that could be used.
>>>
>>>                 How does one specify different rules for 8 digit
>>>                 numbers and 10 digit numbers.
>>>
>>>
>>>             drouting uses longest prefix matching to select the rule.
>>>
>>>             Do you have rules only on length or also on prefix? If
>>>             you provide a bit more details about what routing rules
>>>             you are looking for, we may be able to provide hints
>>>             about how to implement in the config file.
>>>
>>>             Cheers,
>>>             Daniel
>>>
>>>             --             Daniel-Constantin Mierla - 
>>> http://www.asipto.com
>>>             http://twitter.com/#!/miconda
>>>             <http://twitter.com/#%21/miconda> -
>>>             http://www.linkedin.com/in/miconda
>>>             Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>>             http://asipto.com/u/kat
>>>
>>>
>>>
>>>
>>>         --         O: 2248-2483
>>>         C: 6040-3624
>>>
>>>
>>
>>         --         Daniel-Constantin Mierla -http://www.asipto.com
>>         http://twitter.com/#!/miconda 
>> <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda
>>         Kamailio Advanced Training, Berlin, Nov 5-8, 2012 
>> -http://asipto.com/u/kat
>>
>>
>>
>>
>>     --     O: 2248-2483
>>     C: 6040-3624
>>
>>
>>
>>
>>
>> -- 
>> O: 2248-2483
>> C: 6040-3624
>>
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Tue Aug 28 08:19:19 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Aug 2012 08:19:19 +0200
Subject: [SR-Users] [ot] virtualization systems
Message-ID: <503C62E7.6010505@gmail.com>

Hello,

just asking to see your experience deploying sip platforms on virtual 
systems. So far I was running Kamailio in virtual machines and no 
problems, but I insisted that media servers to be on physical machines. 
Lately is more pressure from the market to go everything virtual.

So the question is more about having everything on virtual systems, 
proxy and media server, where the media server can deal with 
transcoding, conference rooms and IVRs.

Any strong comments pro or against?

What is your preferred virtualization system for such deployments?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From jh at tutpro.com  Tue Aug 28 08:24:19 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Tue, 28 Aug 2012 09:24:19 +0300
Subject: [SR-Users]  [ot] virtualization systems
In-Reply-To: <503C62E7.6010505@gmail.com>
References: <503C62E7.6010505@gmail.com>
Message-ID: <20540.25619.132684.141948@siika.tutpro.com>

Daniel-Constantin Mierla writes:

> So the question is more about having everything on virtual systems, 
> proxy and media server, where the media server can deal with 
> transcoding, conference rooms and IVRs.

a few years ago i had latency problems when running sems and mediaproxy
in virtual machines (virtualbox), but i'm sure things have improved
since then.

-- juha


From bligthart at btlnet.co.uk  Tue Aug 28 09:21:08 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Tue, 28 Aug 2012 07:21:08 +0000
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503C62E7.6010505@gmail.com>
References: <503C62E7.6010505@gmail.com>
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11E403@DB3PRD0510MB380.eurprd05.prod.outlook.com>

Hi,

When there is no over contention of the physical machines that run your VM machines on all is ok.

However finding/ fixing issues around I/O is made harder by having virtual devices since all types
of I/O are affected.

My preference is the same as yours. 

Regards,

Bou



-----Original Message-----
From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Daniel-Constantin Mierla
Sent: 28 August 2012 07:19
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
Subject: [SR-Users] [ot] virtualization systems

Hello,

just asking to see your experience deploying sip platforms on virtual systems. So far I was running Kamailio in virtual machines and no problems, but I insisted that media servers to be on physical machines. 
Lately is more pressure from the market to go everything virtual.

So the question is more about having everything on virtual systems, proxy and media server, where the media server can deal with transcoding, conference rooms and IVRs.

Any strong comments pro or against?

What is your preferred virtualization system for such deployments?

Cheers,
Daniel

--
Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat


_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




From klaus.mailinglists at pernau.at  Tue Aug 28 09:36:34 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 28 Aug 2012 09:36:34 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <20120824144154.3DAD39EA@centrum.sk>
References: <20120823115419.63CC468C@centrum.sk>, <5036040F.3030908@gmail.com>,
	<20120823135149.427B12C2@centrum.sk> <50362A69.8060807@pernau.at>
	<20120824144154.3DAD39EA@centrum.sk>
Message-ID: <503C7502.2090609@pernau.at>



On 24.08.2012 14:41, martian at centrum.sk wrote:
> The Route and Record-route headers are identical.
>
>
>  From debug (when alias=domain.ch:5060):
>
> ----authentication of INVITE:
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: NOTICE: <script>:
> ---------------------- In route(AUTH), just before
> from_uri==myself ----------------------
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==9 &&
> [domain.ch] == [127.0.0.1]
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==15 &&
> [domain.ch] == [<IP_ADDRESS_OF_KAMAILIO>]
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==9 &&
> [domain.ch] == [127.0.0.1]
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==15 &&
> [domain.ch] == [<IP_ADDRESS_OF_KAMAILIO>]
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: <core>
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: NOTICE: <script>:
> ---------------------- from_uri==myself evaluated as
> TRUE!! ----------------------


Is this really a complete log? According to the log uri==myself should 
return FALSE as the compared strings are never the same.

> When I set alias=server.domain.ch:5060, from_uri==myself returns false
> (when determining if INVITE should be authenticated,resulting in
> replying 100 trying instead of 407 Proxy Auth Req) and loose_route()
> starts returning true and relays the ACK correctly.
>
> I can post more debug from this case also, but I didn't want to spam so
> much in one message.
>
> If you would like to see it, please let me know.
>
> So .. Shall I consider the loose_route() part fixed and assume that
> there MUST be a full name (hostname.domain:port) in the alias, when
> Kamailio is not used as a primary proxy for the domain?

No. It is rather simple: domain.ch is not identical to domain.ch:5060 
(as the first URI results in NAPTR+SRV lookups and my use another port 
than 5060).

Thus, if you want that Kamailio detects domain.ch as local domain, add 
"alias=domain.ch". If you want that Kamailio detects domain.ch:5060 as 
local domain add alias=domain.ch:5060 (not sure if quotes are needed here).

If you want that Kamailio accepts both domains as local domains, then 
add both alias.

Regardind loose_route: As Daniel mentioned, the ACK is broken.

regards
Klaus

>
> What about the from_uri==myself part?
>
> Martin
>
> ______________________________________________________________
>  > Od: "Klaus Darilion" <klaus.mailinglists at pernau.at>
>  > Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List" <sr-users at lists.sip-router.org>
>  > D?tum: 23.08.2012 15:04
>  > Predmet: Re: [SR-Users] Possible bug in authentication
>  >
>
>  > CC: miconda at gmail.com
>
> The Route URI (sent by SBC) must be identical to the Record-Route URI
> (inserted by Kamailio).
>
> To find out why loose_route returns FALSE increase log-level.
> loose_route uses the "ismyself" function to evaluate if the Route header
> addresses this Kamailio server. And the "ismyself" is very verbose when
> doing this check.
>
> regards
> Klaus
>
> On 23.08.2012 13:51, martian at centrum.sk wrote:
>  > Ok, so .. I have a session border controller device that is a contact
>  > point for my SIP domain (SRV record in DNS set to its IP). All the
>  > trafic goes through it and it does things like topology hiding etc.. The
>  > device forwards the INVITE messages to Kamailio, because of the routing.
>  >
>  > The loose_route was working strangely, because it did not behave as
>  > described in the documentation.
>  >
>  > Here is the sip message that it was suppose to pass:
>  >
>  > ACK sip:acc1 at domain.ch:5060 SIP/2.0
>  >
>  > Via: SIP/2.0/UDP domain.ch;branch=z9hG4bKac386033013
>  >
>  > Max-Forwards: 70
>  >
>  > From: "acc2" <sip:acc2 at domain.ch>;tag=1c1749458918
>  >
>  > To: <sip:acc1@<IP_ADRESS_OF_KAMAILIO>;user=phone>;tag=1c1892801634
>  >
>  > Call-ID: 17494024742382012111116@<IP_ADDRESS_OF_SBC>
>  >
>  > CSeq: 2 ACK
>  >
>  > Contact: <sip:acc2 at domain.ch:5060>
>  >
>  > Route: <sip:<IP_ADDRESS_OF_KAMAILIO>;lr=on>
>  >
>  > Supported: em,timer,replaces,path,resource-priority
>  >
>  > Allow:
>  >
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
>  >
>  > User-Agent: SBC_DEVICE
>  >
>  > Content-Length: 0
>  >
>  > As you can see, there is a Route header and a To_tag .. so the
>  > loose_route function should return true. But instead, it returned false,
>  > then t_check_trans() also returned false and the routing logic exited
>  > (exit;).
>  >
>  > This happens when the value of alias is not enclosed in double quotes.
>  >
>  > PS.: There is a "-" symbol in the domain name. Can't that be a problem
>  > causing the need for the double quotes?
>  >
>  > PS2: Should there be only a domain name in the alias? or also the
>  > hostname part? ... for example:   domain.ch:5060 or server.domain.ch:5060
>  >
>  > Martin
>  >
>  > ______________________________________________________________
>  >  > Od: "Daniel-Constantin Mierla" <miconda at gmail.com>
>  >  > Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>  > Users Mailing List" <sr-users at lists.sip-router.org>
>  >  > D?tum: 23.08.2012 12:21
>  >  > Predmet: Re: [SR-Users] Possible bug in authentication
>  >  >
>  >
>  > Hello,
>  >
>  > On 8/23/12 11:54 AM, martian at centrum.sk <mailto:martian at centrum.sk>
> wrote:
>  >
>  >     Hello to everybody.
>  >
>  >     I am currently working with Kamailio 3.3.1 on RedHat.
>  >
>  >     The "loose_route" function was not working correctly and I observed
>  >     some very strange behaviour (not as one described in the
>  >     documentation of the function).
>  >
>  >     I have found that there needs to be a port included in the "alias"
>  >     variable for the loose_route function to work correctly.
>  >
>  >     However, upon adding the port to alias, the INVITE messages were no
>  >     longer authenticated (Kamailio just accepted them and didn't send
>  >     proxy-auth header in 407 message).
>  >
>  >     My alias:
>  >
>  >     alias="domain.ch:5060"
>  >
>  >     Examining default routing logic, I found the problem here:
>  >
>  >     if (is_method("REGISTER") || from_uri==myself)
>  >
>  >     {
>  >
>  >     # authenticate requests
>  >
>  >     ...
>  >
>  >     }
>  >
>  >     The "from_uri==myself" was no longer evaluated as true, because
>  >     there was a port at the end of the alias.
>  >
>  >     The FROM Header of the INVITE messages looks like:
>  >
>  >     From: "acc1" <sip:acc1 at domain.ch>;tag=12345
>  >
>  >     ..so .. no port number there.
>  >
>  >     Btw, I have fixed this with replacing the "myself" list with my own
>  >     defined variable MY_DOMAIN.
>  >
>  >     #!define MY_DOMAIN ".*@domain.ch" <mailto:.*@domain.ch>
>  >
>  >     So now the condition looks like this:
>  >
>  >     if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
>  >
>  >     {
>  >
>  >     ...
>  >
>  >     }
>  >
>  >     I am not sure if this is a bug that needs to be fixed or not. I am
>  >     just pointing my finger at it and I hope it will contribute to the
>  >     development.
>  >
>  >     Also, a valid description of this behavior (when using port in
>  >     alias) would be appreciated.
>  >
>  >
>  > if you enclose the value of the alias parameter in double quotes, then
>  > it is taken as string value. If you want to set it to a host:port, then
>  > remove the double quotes:
>  >
>  > alias=domain.ch:5060
>  >
>  >
>  > Why do you say the loose_route() was working strangely? Do you add the
>  > hostname as record-route, not the IP address? Detail more about what you
>  > think is wrong with record routing/loose routing.
>  >
>  >
>  > Cheers,
>  > Daniel
>  >
>  > -- Daniel-Constantin Mierla
> -http://www.asipto.comhttp://twitter.com/#
> <http://www.asipto.comhttp//twitter.com/>!/miconda
>   -http://www.linkedin.com/in/micondaKamailio Advanced Training, Berlin,
> Nov 5-8, 2012 -http://asipto.com/u/kat
>  >
>  >
>  >
>  > _______________________________________________
>  > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>  > sr-users at lists.sip-router.org
>  > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>  >
>


From carsten at ng-voice.com  Tue Aug 28 09:59:04 2012
From: carsten at ng-voice.com (Carsten Bock)
Date: Tue, 28 Aug 2012 09:59:04 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503C62E7.6010505@gmail.com>
References: <503C62E7.6010505@gmail.com>
Message-ID: <CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>

Hi Daniel,

here's from my personal experience:
Our setup at ng-voice is a little weird sometimes: We've rented some
virtual servers at a german provider (who uses Xen). On these virtual
servers we've installed OpenVz, which for us is absolutely great, if
you are just working with Linux-Servers. While Xen is a rather
complete virtualization, OpenVz is lightweight and comes in handy, if
you just want to logically separate servers. We've got each IMS
component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
on a dedicated OpenVz Container, which is really great. We've even got
a CentOs-Container running on a Debian OpenVz, which is started
"on-demand" in order to build RPM-Packages. With OpenVz you can even
move Containers from one host to another, theoretically with zero
downtime (doesn't work with SEMS, don't know about other software).
For our IMS-setup, we work with RTP-Relaying, which works great within
virtualization, i cannot complain.

At another customer (a fibre provider in Germany), we're running all
the infrastructure on Xen-only. An infrastructure provider takes care
of the administration, but those servers run poorly (RTP-Relaying is
okay but everything else is really slow).

Conclusion for me: VoIP on virtual servers can work great, but the
virtualization infrastructure needs to be administered properly which
may not be an easy task, if you are new in this subject.

Kind regards,
Carsten

2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com>:
> Hello,
>
> just asking to see your experience deploying sip platforms on virtual
> systems. So far I was running Kamailio in virtual machines and no problems,
> but I insisted that media servers to be on physical machines. Lately is more
> pressure from the market to go everything virtual.
>
> So the question is more about having everything on virtual systems, proxy
> and media server, where the media server can deal with transcoding,
> conference rooms and IVRs.
>
> Any strong comments pro or against?
>
> What is your preferred virtualization system for such deployments?
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
Carsten Bock
CEO (Gesch?ftsf?hrer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:carsten at ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Gesch?ftsf?hrer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/


From jason.penton at smilecoms.com  Tue Aug 28 10:33:43 2012
From: jason.penton at smilecoms.com (Jason Penton)
Date: Tue, 28 Aug 2012 10:33:43 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
Message-ID: <CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>

Hey Daniel,

We use Solaris virtualisation and it works great. The zones (VMs per se)
are lightweight, easy to administer and rock solid.

btw, common misconceptions are that you need sun (oracle) hardware and that
the os is not free. These are both false.

cheers
Jason

On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock <carsten at ng-voice.com> wrote:

> Hi Daniel,
>
> here's from my personal experience:
> Our setup at ng-voice is a little weird sometimes: We've rented some
> virtual servers at a german provider (who uses Xen). On these virtual
> servers we've installed OpenVz, which for us is absolutely great, if
> you are just working with Linux-Servers. While Xen is a rather
> complete virtualization, OpenVz is lightweight and comes in handy, if
> you just want to logically separate servers. We've got each IMS
> component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
> on a dedicated OpenVz Container, which is really great. We've even got
> a CentOs-Container running on a Debian OpenVz, which is started
> "on-demand" in order to build RPM-Packages. With OpenVz you can even
> move Containers from one host to another, theoretically with zero
> downtime (doesn't work with SEMS, don't know about other software).
> For our IMS-setup, we work with RTP-Relaying, which works great within
> virtualization, i cannot complain.
>
> At another customer (a fibre provider in Germany), we're running all
> the infrastructure on Xen-only. An infrastructure provider takes care
> of the administration, but those servers run poorly (RTP-Relaying is
> okay but everything else is really slow).
>
> Conclusion for me: VoIP on virtual servers can work great, but the
> virtualization infrastructure needs to be administered properly which
> may not be an easy task, if you are new in this subject.
>
> Kind regards,
> Carsten
>
> 2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com>:
> > Hello,
> >
> > just asking to see your experience deploying sip platforms on virtual
> > systems. So far I was running Kamailio in virtual machines and no
> problems,
> > but I insisted that media servers to be on physical machines. Lately is
> more
> > pressure from the market to go everything virtual.
> >
> > So the question is more about having everything on virtual systems, proxy
> > and media server, where the media server can deal with transcoding,
> > conference rooms and IVRs.
> >
> > Any strong comments pro or against?
> >
> > What is your preferred virtualization system for such deployments?
> >
> > Cheers,
> > Daniel
> >
> > --
> > Daniel-Constantin Mierla - http://www.asipto.com
> > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
> http://asipto.com/u/kat
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Carsten Bock
> CEO (Gesch?ftsf?hrer)
>
> ng-voice GmbH
> Schomburgstr. 80
> D-22767 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:carsten at ng-voice.com
>
> Office +49 40 34927219
> Fax +49 40 34927220
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Gesch?ftsf?hrer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>

This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer

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From vitalik.voip at gmail.com  Tue Aug 28 11:19:38 2012
From: vitalik.voip at gmail.com (Vitaliy Aleksandrov)
Date: Tue, 28 Aug 2012 12:19:38 +0300
Subject: [SR-Users] AVP string or integer ID
Message-ID: <503C8D2A.5040805@gmail.com>

Hi all.

I'm working with kamailio for about 1 year and still don't understand 
difference between integer and string IDs of AVPs.
Is it just a question of usability ?



From oej at edvina.net  Tue Aug 28 11:36:26 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Tue, 28 Aug 2012 11:36:26 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
Message-ID: <A511FA0B-049A-46D3-9A44-34BE17D5F0EE@edvina.net>

I haven't heard anyone using Asterisk in large production systems in other virtualization than OpenVZ. Asterisk depends a lot on timers and these aren't reliable enough in Vmware and Xen when putting load on the media server.

I would love to hear about a successful implementation :-) on other platforms but so far I haven't. While waiting, OpenVZ works great.

/O




From klaus.mailinglists at pernau.at  Tue Aug 28 12:11:47 2012
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 28 Aug 2012 12:11:47 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
Message-ID: <503C9963.3080804@pernau.at>

Jason, are you using Solaris only as host or also as guest OS?

regards
Klaus

On 28.08.2012 10:33, Jason Penton wrote:
> Hey Daniel,
>
> We use Solaris virtualisation and it works great. The zones (VMs per se)
> are lightweight, easy to administer and rock solid.
>
> btw, common misconceptions are that you need sun (oracle) hardware and
> that the os is not free. These are both false.
>
> cheers
> Jason
>
> On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock <carsten at ng-voice.com
> <mailto:carsten at ng-voice.com>> wrote:
>
>     Hi Daniel,
>
>     here's from my personal experience:
>     Our setup at ng-voice is a little weird sometimes: We've rented some
>     virtual servers at a german provider (who uses Xen). On these virtual
>     servers we've installed OpenVz, which for us is absolutely great, if
>     you are just working with Linux-Servers. While Xen is a rather
>     complete virtualization, OpenVz is lightweight and comes in handy, if
>     you just want to logically separate servers. We've got each IMS
>     component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
>     on a dedicated OpenVz Container, which is really great. We've even got
>     a CentOs-Container running on a Debian OpenVz, which is started
>     "on-demand" in order to build RPM-Packages. With OpenVz you can even
>     move Containers from one host to another, theoretically with zero
>     downtime (doesn't work with SEMS, don't know about other software).
>     For our IMS-setup, we work with RTP-Relaying, which works great within
>     virtualization, i cannot complain.
>
>     At another customer (a fibre provider in Germany), we're running all
>     the infrastructure on Xen-only. An infrastructure provider takes care
>     of the administration, but those servers run poorly (RTP-Relaying is
>     okay but everything else is really slow).
>
>     Conclusion for me: VoIP on virtual servers can work great, but the
>     virtualization infrastructure needs to be administered properly which
>     may not be an easy task, if you are new in this subject.
>
>     Kind regards,
>     Carsten
>
>     2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com
>     <mailto:miconda at gmail.com>>:
>      > Hello,
>      >
>      > just asking to see your experience deploying sip platforms on virtual
>      > systems. So far I was running Kamailio in virtual machines and no
>     problems,
>      > but I insisted that media servers to be on physical machines.
>     Lately is more
>      > pressure from the market to go everything virtual.
>      >
>      > So the question is more about having everything on virtual
>     systems, proxy
>      > and media server, where the media server can deal with transcoding,
>      > conference rooms and IVRs.
>      >
>      > Any strong comments pro or against?
>      >
>      > What is your preferred virtualization system for such deployments?
>      >
>      > Cheers,
>      > Daniel
>      >
>      > --
>      > Daniel-Constantin Mierla - http://www.asipto.com
>      > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
>     http://www.linkedin.com/in/miconda
>      > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>     http://asipto.com/u/kat
>      >
>      >
>      > _______________________________________________
>      > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>     mailing list
>      > sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>      > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>     --
>     Carsten Bock
>     CEO (Gesch?ftsf?hrer)
>
>     ng-voice GmbH
>     Schomburgstr. 80
>     D-22767 Hamburg / Germany
>
>     http://www.ng-voice.com
>     mailto:carsten at ng-voice.com <mailto:carsten at ng-voice.com>
>
>     Office +49 40 34927219 <tel:%2B49%2040%2034927219>
>     Fax +49 40 34927220 <tel:%2B49%2040%2034927220>
>
>     Sitz der Gesellschaft: Hamburg
>     Registergericht: Amtsgericht Hamburg, HRB 120189
>     Gesch?ftsf?hrer: Carsten Bock
>     Ust-ID: DE279344284
>
>     Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>     http://www.ng-voice.com/imprint/
>
>     _______________________________________________
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>


From fred at teamforrest.com  Tue Aug 28 12:18:19 2012
From: fred at teamforrest.com (Fred Posner)
Date: Tue, 28 Aug 2012 06:18:19 -0400
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503C62E7.6010505@gmail.com>
References: <503C62E7.6010505@gmail.com>
Message-ID: <1F2F944B-1B1E-46CA-9373-8B824AB55517@teamforrest.com>

Same preference-- especially when call load gets high or there is conferencing / call recording. 

---Fred 


On Aug 28, 2012, at 2:19 AM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hello,
> 
> just asking to see your experience deploying sip platforms on virtual systems. So far I was running Kamailio in virtual machines and no problems, but I insisted that media servers to be on physical machines. Lately is more pressure from the market to go everything virtual.
> 
> So the question is more about having everything on virtual systems, proxy and media server, where the media server can deal with transcoding, conference rooms and IVRs.
> 
> Any strong comments pro or against?
> 
> What is your preferred virtualization system for such deployments?
> 
> Cheers,
> Daniel
> 
> -- 
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


From evilzluk at gmail.com  Tue Aug 28 12:31:15 2012
From: evilzluk at gmail.com (Konstantin M.)
Date: Tue, 28 Aug 2012 13:31:15 +0300
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <A511FA0B-049A-46D3-9A44-34BE17D5F0EE@edvina.net>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<A511FA0B-049A-46D3-9A44-34BE17D5F0EE@edvina.net>
Message-ID: <CAEYtTzfmkJKPNHtygKTnvqSq4vwwTK2pHpGDTQR4jJkpVjVUAA@mail.gmail.com>


From jason.penton at smilecoms.com  Tue Aug 28 12:33:57 2012
From: jason.penton at smilecoms.com (Jason Penton)
Date: Tue, 28 Aug 2012 12:33:57 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503C9963.3080804@pernau.at>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
	<503C9963.3080804@pernau.at>
Message-ID: <CAE=KcrgQU+uausY68Y1-fzziqsF6O60OtGCX-W1W7P0s-EbLzA@mail.gmail.com>

Hey Klaus,

We use solaris as host AND guest ;)

solaris is slightly different to normal VM systems in that you can't really
load up "any OS" into the guest. There are branded zones that support a few
flavours of linux (ala redhat, etc). However, in our experience the Solaris
OS is really solid for all we need.

you can read up alot here -
http://www.oracle.com/us/products/servers-storage/solaris/solaris11/overview/index.html

Also there is some really nice network virtualisation you can do too.

Cheers
Jason



On Tue, Aug 28, 2012 at 12:11 PM, Klaus Darilion <
klaus.mailinglists at pernau.at> wrote:

> Jason, are you using Solaris only as host or also as guest OS?
>
> regards
> Klaus
>
>
> On 28.08.2012 10:33, Jason Penton wrote:
>
>> Hey Daniel,
>>
>> We use Solaris virtualisation and it works great. The zones (VMs per se)
>> are lightweight, easy to administer and rock solid.
>>
>> btw, common misconceptions are that you need sun (oracle) hardware and
>> that the os is not free. These are both false.
>>
>> cheers
>> Jason
>>
>> On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock <carsten at ng-voice.com
>> <mailto:carsten at ng-voice.com>> wrote:
>>
>>     Hi Daniel,
>>
>>     here's from my personal experience:
>>     Our setup at ng-voice is a little weird sometimes: We've rented some
>>     virtual servers at a german provider (who uses Xen). On these virtual
>>     servers we've installed OpenVz, which for us is absolutely great, if
>>     you are just working with Linux-Servers. While Xen is a rather
>>     complete virtualization, OpenVz is lightweight and comes in handy, if
>>     you just want to logically separate servers. We've got each IMS
>>     component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
>>     on a dedicated OpenVz Container, which is really great. We've even got
>>     a CentOs-Container running on a Debian OpenVz, which is started
>>     "on-demand" in order to build RPM-Packages. With OpenVz you can even
>>     move Containers from one host to another, theoretically with zero
>>     downtime (doesn't work with SEMS, don't know about other software).
>>     For our IMS-setup, we work with RTP-Relaying, which works great within
>>     virtualization, i cannot complain.
>>
>>     At another customer (a fibre provider in Germany), we're running all
>>     the infrastructure on Xen-only. An infrastructure provider takes care
>>     of the administration, but those servers run poorly (RTP-Relaying is
>>     okay but everything else is really slow).
>>
>>     Conclusion for me: VoIP on virtual servers can work great, but the
>>     virtualization infrastructure needs to be administered properly which
>>     may not be an easy task, if you are new in this subject.
>>
>>     Kind regards,
>>     Carsten
>>
>>     2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com
>>     <mailto:miconda at gmail.com>>:
>>
>>      > Hello,
>>      >
>>      > just asking to see your experience deploying sip platforms on
>> virtual
>>      > systems. So far I was running Kamailio in virtual machines and no
>>     problems,
>>      > but I insisted that media servers to be on physical machines.
>>     Lately is more
>>      > pressure from the market to go everything virtual.
>>      >
>>      > So the question is more about having everything on virtual
>>     systems, proxy
>>      > and media server, where the media server can deal with transcoding,
>>      > conference rooms and IVRs.
>>      >
>>      > Any strong comments pro or against?
>>      >
>>      > What is your preferred virtualization system for such deployments?
>>      >
>>      > Cheers,
>>      > Daniel
>>      >
>>      > --
>>      > Daniel-Constantin Mierla - http://www.asipto.com
>>      > http://twitter.com/#!/miconda <http://twitter.com/#%21/**miconda<http://twitter.com/#%21/miconda>>
>> -
>>
>>     http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda>
>>      > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>     http://asipto.com/u/kat
>>      >
>>      >
>>      > ______________________________**_________________
>>      > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>     mailing list
>>      > sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-**
>> router.org <sr-users at lists.sip-router.org>>
>>
>>      > http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
>>
>>     --
>>     Carsten Bock
>>     CEO (Gesch?ftsf?hrer)
>>
>>     ng-voice GmbH
>>     Schomburgstr. 80
>>     D-22767 Hamburg / Germany
>>
>>     http://www.ng-voice.com
>>     mailto:carsten at ng-voice.com <mailto:carsten at ng-voice.com>
>>
>>     Office +49 40 34927219 <tel:%2B49%2040%2034927219>
>>     Fax +49 40 34927220 <tel:%2B49%2040%2034927220>
>>
>>
>>     Sitz der Gesellschaft: Hamburg
>>     Registergericht: Amtsgericht Hamburg, HRB 120189
>>     Gesch?ftsf?hrer: Carsten Bock
>>     Ust-ID: DE279344284
>>
>>     Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>>     http://www.ng-voice.com/**imprint/ <http://www.ng-voice.com/imprint/>
>>
>>     ______________________________**_________________
>>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list
>>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-**router.org<sr-users at lists.sip-router.org>
>> >
>>     http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
>> This email is subject to the disclaimer of Smile Communications (PTY)
>> Ltd. at http://www.smilecoms.com/**disclaimer<http://www.smilecoms.com/disclaimer>
>>
>>
>>
>>
>>
>> ______________________________**_________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
>>
>>

This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer

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From phillman25 at gmail.com  Tue Aug 28 13:26:38 2012
From: phillman25 at gmail.com (phillman25)
Date: Tue, 28 Aug 2012 14:26:38 +0300
Subject: [SR-Users] Trunk Utilization graphing using MI COMMANDS
In-Reply-To: <503BD63C.9010809@gmail.com>
References: <CAEprJzJ35sRgTmwrwpGO36JqHHbpSYZac_aZvO8D0QbXfWoomg@mail.gmail.com>
	<503B1AA5.5090505@gmail.com>
	<CAEprJzJzr-qGKh138V866=e3z5GAMMJBdJFipypWGsgiAe+10A@mail.gmail.com>
	<503B472E.7010603@gmail.com>
	<CAEprJzL6hM_L_Y3dBk3NpMzaOF-XWuE05b77rG5+M3Rkp0GsFg@mail.gmail.com>
	<503BD63C.9010809@gmail.com>
Message-ID: <CAEprJzLX=FUh1pR7NMbguFf6drrBsfGWeyFdGWA72BAdh-hqhA@mail.gmail.com>

Hi Daniel

Sure i will prepare something and let you know.

Just brief info:
For current number of dialogs i used SNMPSTATS module on Kamailio, and use
snmpget function from a remote server (CACTI) to poll data i.e.

snmpget -v2c -c public xx.xx.xx.xx .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 | cut
-b58-65

where .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 corresponds to MIB
(openserCurNumDialogs.0)  and xx.xx.xx.xx is kamailio ip.

I then monitor calls per provider using dialog module and profile_get_size
function together with sercmd and ctl module to poll data into CACTI i.e.

sudo sercmd -s udp:xx.xx.xx.xx:3012 mi profile_get_size PROFILE WITH VALUE

I will write what i did in more detail though.

Regards
Phillip


On Mon, Aug 27, 2012 at 11:19 PM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

>  Hello,
>
>
> On 8/27/12 4:06 PM, phillman25 wrote:
>
> Thanks Daniel i was able to implement what you advised below and am able
> to poll specific data into CACTI successfully.
>
> great! Have you made it like a plugin? Maybe you can write few words about
> or make web page available for it, there is a similar plugin for munin,
> published by Inaki:
>
> - http://dev.sipdoc.net/projects/sip-stuff/wiki/KamailioMuninPlugin
>
> Cheers,
> Daniel
>
>
>
>  Thanks again!
> Phillip
>
>
> On Mon, Aug 27, 2012 at 1:08 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> I am not sure it supports right now sending from the same socket, but it
>> should send back to same ip and port from where it received the command. A
>> patch it might be required to get it like that, so far I don't remember
>> anyone willing to get such thing.
>>
>> Btw, another alternative, you can eventually use sercmd with ctl module
>> -- it works also over tcp (as well as udp). You can run any MI command via
>> 'mi', like:
>>
>> sercmd ... mi address_dump
>>
>> Look at sercmd -h for the options.
>>
>> So you don't need to build a script around nc.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/27/12 11:21 AM, phillman25 wrote:
>>
>> Thanks for the info Daniel.
>>
>>  I have implemented mi_datagram module and defined it with the following
>> parameters:
>>
>>  modparam("mi_datagram", "socket_name", "udp:2.2.2.2:8033")
>>
>>  When i run the below command from a remote server:
>>
>>  echo -ne ":address_dump:\n\n" | nc -w 1 -u 2.2.2.2 8033
>>
>>  i don't receive an output, but when i monitor all traffic on the local
>> machine:
>>
>>  ngrep -q -t -P "" -W byline -d any host 2.2.2.2 and udp
>>
>>  i observe the output:
>>
>>  U 2012/08/27 12:41:36.735094 1.1.1.1:55621 -> 2.2.2.2:8033
>> :address_dump:
>>
>>
>>  U 2012/08/27 12:41:36.735201 2.2.2.2:47024 -> 1.1.1.1:55621
>> 200 OK
>>   71 <1, xx.xx.xx.xx, 0> [1]
>>   72 <1, yy.yy.yy.yy, 0> [4]
>>
>>
>>  Seems like the command is accepted from remote server and executed on
>> local machine but not relayed back to the remote server. Looks like
>> mi_datagram replies on different port other than 8033. Is there a way to
>> force the reply on source port 8033?
>>
>>  Thanks
>> Phillip
>>
>>
>>
>>
>>
>>
>> On Mon, Aug 27, 2012 at 9:58 AM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> if you want to execute from a remote server, maybe it is better to use
>>> mi_datagram or xmlrpc modules to run MI/RPC commands.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>> On 8/22/12 12:02 PM, phillman25 wrote:
>>>
>>>  Dear List
>>>
>>>  I am trying to run the below command from a REMOTE server where i use
>>> Cacti to graph all my data. I am trying to graph on a per trunk basis.
>>>
>>>  kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
>>>
>>>  This command yields an output on the local server.
>>>
>>>  Could someone perhaps point me in the right direction?
>>>
>>>
>>>  Thanks!
>>>
>>>
>>>  _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>
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From oej at edvina.net  Tue Aug 28 14:56:53 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Tue, 28 Aug 2012 14:56:53 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <CAEYtTzfmkJKPNHtygKTnvqSq4vwwTK2pHpGDTQR4jJkpVjVUAA@mail.gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<A511FA0B-049A-46D3-9A44-34BE17D5F0EE@edvina.net>
	<CAEYtTzfmkJKPNHtygKTnvqSq4vwwTK2pHpGDTQR4jJkpVjVUAA@mail.gmail.com>
Message-ID: <76475D74-197E-4E8A-A980-1F29848B43AA@edvina.net>


28 aug 2012 kl. 12:31 skrev "Konstantin M." <evilzluk at gmail.com>:

> >From my previous experience, I can't say, it's a correct.
> I'm using asterisk (over 3 years) and kamailio (about 1 year) in a *production* environment using Xen and I have almost no problems with timers (they were previously with 100Hz),
> but I have experienced A LOT of problems with asterisk or any voip-related projects using OpenVZ.
> So from my own perspective, OpenVZ (which is a more cheaper) is only suitable for hosting (web or such of tasks) or any non-realtime tasks.
> Maybe it was caused by incorrect settings on host (which was out of my control).
> Maybe I'm wrong but to me, Xen is better vs OpenVZ :-)
That's weird. I've been using OpenVZ in productions sites handling 1500 concurrent calls without any issues. For years.

I guess it's a matter of personal preferences. Good to hear that you have it working in Xen!

/O
> 
> 
> 2012/8/28 Olle E. Johansson <oej at edvina.net>
> I haven't heard anyone using Asterisk in large production systems in other virtualization than OpenVZ. Asterisk depends a lot on timers and these aren't reliable enough in Vmware and Xen when putting load on the media server.
> 
> I would love to hear about a successful implementation :-) on other platforms but so far I haven't. While waiting, OpenVZ works great.
> 
> /O
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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From bligthart at btlnet.co.uk  Tue Aug 28 16:09:54 2012
From: bligthart at btlnet.co.uk (Boudewyn Ligthart)
Date: Tue, 28 Aug 2012 14:09:54 +0000
Subject: [SR-Users] Cassandra module - query question
Message-ID: <D8261FE064C78B44B77EF7C4569D6CBF0D11E58B@DB3PRD0510MB380.eurprd05.prod.outlook.com>

Hi,

I would to query the Cassandra database for other information with either CQL or other commands.
Could this be done through means of either avpops or sqlops, like an MySQL database.

This could be done through an external script but I would like to utilize the connection that is already
established.

Thanks,

Bou


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From miconda at gmail.com  Tue Aug 28 17:48:58 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Aug 2012 17:48:58 +0200
Subject: [SR-Users] Cassandra module - query question
In-Reply-To: <D8261FE064C78B44B77EF7C4569D6CBF0D11E58B@DB3PRD0510MB380.eurprd05.prod.outlook.com>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11E58B@DB3PRD0510MB380.eurprd05.prod.outlook.com>
Message-ID: <503CE86A.30900@gmail.com>

Hello,

I am not familiar with the insights of cassandra module, but sqlops is 
designed to deal with raw sql queries, so it might not work at all with 
cassandra module.

Using avpops could work with avp_db_load(), because that function is 
using the internal DB API INSERT which is in use by generic insert 
operations, like storing the location records. But avp_db_query() 
requires again sql query access, so it might not work.

Cheers,
Daniel

On 8/28/12 4:09 PM, Boudewyn Ligthart wrote:
>
> Hi,
>
> I would to query the Cassandra database for other information with 
> either CQL or other commands.
> Could this be done through means of either avpops or sqlops, like an 
> MySQL database.
>
> This could be done through an external script but I would like to 
> utilize the connection that is already
> established.
>
> Thanks,
>
> Bou
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Tue Aug 28 17:52:47 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Aug 2012 17:52:47 +0200
Subject: [SR-Users] AVP string or integer ID
In-Reply-To: <503C8D2A.5040805@gmail.com>
References: <503C8D2A.5040805@gmail.com>
Message-ID: <503CE94F.1020107@gmail.com>

Hello,

On 8/28/12 11:19 AM, Vitaliy Aleksandrov wrote:
> Hi all.
>
> I'm working with kamailio for about 1 year and still don't understand 
> difference between integer and string IDs of AVPs.
> Is it just a question of usability ?
yes, it is just a matter of addressing, integers id were supposed to be 
faster to find in the AVPs list, but they are more meaningless, making 
them harder to work with over the time. String IDs have also internal 
optimizations for searching, so there is not noticeable performance 
difference.

It is recommended to use strings, unless one is really concerned about 
performance.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Tue Aug 28 17:56:22 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Aug 2012 17:56:22 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
Message-ID: <503CEA26.4000606@gmail.com>

Hello,

On 8/28/12 10:33 AM, Jason Penton wrote:
> Hey Daniel,
>
> We use Solaris virtualisation and it works great. The zones (VMs per 
> se) are lightweight, easy to administer and rock solid.
>
> btw, common misconceptions are that you need sun (oracle) hardware and 
> that the os is not free. These are both false.

so you use opensolaris, I guess, and then it can be any intel/amd arch 
server (e.g., dell, hp)?

Cheers,
Daniel

>
> cheers
> Jason
>
> On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock <carsten at ng-voice.com 
> <mailto:carsten at ng-voice.com>> wrote:
>
>     Hi Daniel,
>
>     here's from my personal experience:
>     Our setup at ng-voice is a little weird sometimes: We've rented some
>     virtual servers at a german provider (who uses Xen). On these virtual
>     servers we've installed OpenVz, which for us is absolutely great, if
>     you are just working with Linux-Servers. While Xen is a rather
>     complete virtualization, OpenVz is lightweight and comes in handy, if
>     you just want to logically separate servers. We've got each IMS
>     component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
>     on a dedicated OpenVz Container, which is really great. We've even got
>     a CentOs-Container running on a Debian OpenVz, which is started
>     "on-demand" in order to build RPM-Packages. With OpenVz you can even
>     move Containers from one host to another, theoretically with zero
>     downtime (doesn't work with SEMS, don't know about other software).
>     For our IMS-setup, we work with RTP-Relaying, which works great within
>     virtualization, i cannot complain.
>
>     At another customer (a fibre provider in Germany), we're running all
>     the infrastructure on Xen-only. An infrastructure provider takes care
>     of the administration, but those servers run poorly (RTP-Relaying is
>     okay but everything else is really slow).
>
>     Conclusion for me: VoIP on virtual servers can work great, but the
>     virtualization infrastructure needs to be administered properly which
>     may not be an easy task, if you are new in this subject.
>
>     Kind regards,
>     Carsten
>
>     2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com
>     <mailto:miconda at gmail.com>>:
>     > Hello,
>     >
>     > just asking to see your experience deploying sip platforms on
>     virtual
>     > systems. So far I was running Kamailio in virtual machines and
>     no problems,
>     > but I insisted that media servers to be on physical machines.
>     Lately is more
>     > pressure from the market to go everything virtual.
>     >
>     > So the question is more about having everything on virtual
>     systems, proxy
>     > and media server, where the media server can deal with transcoding,
>     > conference rooms and IVRs.
>     >
>     > Any strong comments pro or against?
>     >
>     > What is your preferred virtualization system for such deployments?
>     >
>     > Cheers,
>     > Daniel
>     >
>     > --
>     > Daniel-Constantin Mierla - http://www.asipto.com
>     > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda>
>     - http://www.linkedin.com/in/miconda
>     > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>     http://asipto.com/u/kat
>     >
>     >
>     > _______________________________________________
>     > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>     mailing list
>     > sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>     > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>     --
>     Carsten Bock
>     CEO (Gesch?ftsf?hrer)
>
>     ng-voice GmbH
>     Schomburgstr. 80
>     D-22767 Hamburg / Germany
>
>     http://www.ng-voice.com
>     mailto:carsten at ng-voice.com <mailto:carsten at ng-voice.com>
>
>     Office +49 40 34927219 <tel:%2B49%2040%2034927219>
>     Fax +49 40 34927220 <tel:%2B49%2040%2034927220>
>
>     Sitz der Gesellschaft: Hamburg
>     Registergericht: Amtsgericht Hamburg, HRB 120189
>     Gesch?ftsf?hrer: Carsten Bock
>     Ust-ID: DE279344284
>
>     Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>     http://www.ng-voice.com/imprint/
>
>     _______________________________________________
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>     list
>     sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org>
>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer
>
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From jason.penton at smilecoms.com  Tue Aug 28 18:09:15 2012
From: jason.penton at smilecoms.com (Jason Penton)
Date: Tue, 28 Aug 2012 18:09:15 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503CEA26.4000606@gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
	<503CEA26.4000606@gmail.com>
Message-ID: <CAE=KcrjJJT-esq65aoTTNJApPxTv9EPrFt9dzzzd9uUy_bpeFA@mail.gmail.com>

No. We use solaris11. But yes on any hardware
On Aug 28, 2012 5:56 PM, "Daniel-Constantin Mierla" <miconda at gmail.com>
wrote:

>  Hello,
>
> On 8/28/12 10:33 AM, Jason Penton wrote:
>
> Hey Daniel,
>
> We use Solaris virtualisation and it works great. The zones (VMs per se)
> are lightweight, easy to administer and rock solid.
>
> btw, common misconceptions are that you need sun (oracle) hardware and
> that the os is not free. These are both false.
>
>
> so you use opensolaris, I guess, and then it can be any intel/amd arch
> server (e.g., dell, hp)?
>
> Cheers,
> Daniel
>
>
> cheers
> Jason
>
> On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock <carsten at ng-voice.com>wrote:
>
>> Hi Daniel,
>>
>> here's from my personal experience:
>> Our setup at ng-voice is a little weird sometimes: We've rented some
>> virtual servers at a german provider (who uses Xen). On these virtual
>> servers we've installed OpenVz, which for us is absolutely great, if
>> you are just working with Linux-Servers. While Xen is a rather
>> complete virtualization, OpenVz is lightweight and comes in handy, if
>> you just want to logically separate servers. We've got each IMS
>> component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
>> on a dedicated OpenVz Container, which is really great. We've even got
>> a CentOs-Container running on a Debian OpenVz, which is started
>> "on-demand" in order to build RPM-Packages. With OpenVz you can even
>> move Containers from one host to another, theoretically with zero
>> downtime (doesn't work with SEMS, don't know about other software).
>> For our IMS-setup, we work with RTP-Relaying, which works great within
>> virtualization, i cannot complain.
>>
>> At another customer (a fibre provider in Germany), we're running all
>> the infrastructure on Xen-only. An infrastructure provider takes care
>> of the administration, but those servers run poorly (RTP-Relaying is
>> okay but everything else is really slow).
>>
>> Conclusion for me: VoIP on virtual servers can work great, but the
>> virtualization infrastructure needs to be administered properly which
>> may not be an easy task, if you are new in this subject.
>>
>> Kind regards,
>> Carsten
>>
>> 2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com>:
>>  > Hello,
>> >
>> > just asking to see your experience deploying sip platforms on virtual
>> > systems. So far I was running Kamailio in virtual machines and no
>> problems,
>> > but I insisted that media servers to be on physical machines. Lately is
>> more
>> > pressure from the market to go everything virtual.
>> >
>> > So the question is more about having everything on virtual systems,
>> proxy
>> > and media server, where the media server can deal with transcoding,
>> > conference rooms and IVRs.
>> >
>> > Any strong comments pro or against?
>> >
>> > What is your preferred virtualization system for such deployments?
>> >
>> > Cheers,
>> > Daniel
>> >
>> > --
>> > Daniel-Constantin Mierla - http://www.asipto.com
>> > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>> http://asipto.com/u/kat
>> >
>> >
>> > _______________________________________________
>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> > sr-users at lists.sip-router.org
>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>  --
>> Carsten Bock
>> CEO (Gesch?ftsf?hrer)
>>
>> ng-voice GmbH
>> Schomburgstr. 80
>> D-22767 Hamburg / Germany
>>
>> http://www.ng-voice.com
>> mailto:carsten at ng-voice.com
>>
>> Office +49 40 34927219 <%2B49%2040%2034927219>
>> Fax +49 40 34927220 <%2B49%2040%2034927220>
>>
>> Sitz der Gesellschaft: Hamburg
>> Registergericht: Amtsgericht Hamburg, HRB 120189
>> Gesch?ftsf?hrer: Carsten Bock
>> Ust-ID: DE279344284
>>
>> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>> http://www.ng-voice.com/imprint/
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer
>
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>

This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer

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From oyvind.kolbu at usit.uio.no  Tue Aug 28 18:34:28 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Tue, 28 Aug 2012 18:34:28 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <20120824084809.GI16374@putsch.kolbu.ws>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
Message-ID: <20120828163428.GC75367@putsch.kolbu.ws>

On 2012-08-24 at 10:48, ?yvind Kolbu wrote:
> First impression is that everything is working now, thanks! At least
> registrations are getting populated in both databases from both servers. As
> I'm out traveling I havn't yet tested calling and whether or not the
> failover works as expected. I assume setting usrloc's db_mode to 3 is
> recommended in this kind of setup?
> 
> For now I'm using:
>     modparam("db_cluster", "cluster","data2=>voip1_data2=9s9p;voip2_data2=8s9p")
> on my voip1 machine, and the line below on the voip2 machine:
>     modparam("db_cluster", "cluster","data2=>voip2_data2=9s9p;voip1_data2=8s9p")
> 
> The same configuration created coredumps on both machines previously.

Have played with usrloc and db_cluster a bit in the lab and think I found
two bugs. I have not inspected the actual sql-commands, but this is based
in my observations. Both sql-servers are running postgresql 9.1.4.

1) If the second server lacks an entry in the location table it will never
   be populated, while the primary is updated. This is a major flaw if
   the second server has been unavailable and then is back again it will
   never contain the same information as the primary.

   I think this is because usrloc issues an UPDATE for the row it read,
   but the second server can not UPDATE an non existent row.

   Deleting the entry from the primary server will issue an INSERT to both
   databases and then they are identical again, for that row at least.

2) If the primary lacks an entry and the second already has it, it will
   get an extra entry for that user. After a while though the stale entry
   will be pruned.

   Probably similar reason as above. usrloc finds no entries in the table
   when reading, thus issuing INSERT.

Tried setting

    modparam("usrloc", "db_check_update", 1)

but to no avail.

Even enabling

   modparam("usrloc", "db_update_as_insert", 1)

will not fix issue #1.

-- 
?yvind Kolbu


From ibc at aliax.net  Tue Aug 28 18:46:16 2012
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Tue, 28 Aug 2012 18:46:16 +0200
Subject: [SR-Users] [OT] OverSIP SIP Proxy released
Message-ID: <CALiegfnvMero6t7qTeLGWCC48wAAFdGB8Zf=Kt+SUiH1rOzaBw@mail.gmail.com>

Hi all,

I announce the first stable release of OverSIP SIP Proxy:

  http://www.oversip.net


OverSIP features:
-----------------------------------------------------------------------------

- SIP over UDP, TCP, TLS and WebSocket transports.

- Full support for IPv4, IPv6 and DNS resolution (NAPTR, SRV, A, AAAA).

- Fully programmable using Ruby programming language.

- The perfect Outbound (RFC 5626) Edge Proxy.

- Fully asynchronous event-based design.


OverSIP is not a replacement for well-known proxies as Kamailio
(sip-router). OverSIP does not implement basic proxy features such as
Digest authentication, registration or parallel forking.

OverSIP can behave as an Outbound Edge Proxy between clients and SIP
registrars, or can be placed anywhere for building custom SIP /
WebSocket applications.


Cheers.


-- 
I?aki Baz Castillo
<ibc at aliax.net>


From jmr.richardson at gmail.com  Tue Aug 28 18:51:25 2012
From: jmr.richardson at gmail.com (JR Richardson)
Date: Tue, 28 Aug 2012 11:51:25 -0500
Subject: [SR-Users] [ot] virtualization systems
Message-ID: <CA+U74VOm7=YvnbjwW8X2JFSNwA0oJUf7Pj-=GJxf0Zb2hF4Wdg@mail.gmail.com>

>> just asking to see your experience deploying sip platforms on virtual
>> systems. So far I was running Kamailio in virtual machines and no problems,
>> but I insisted that media servers to be on physical machines. Lately is more
>> pressure from the market to go everything virtual.
>>
>> So the question is more about having everything on virtual systems, proxy
>> and media server, where the media server can deal with transcoding,
>> conference rooms and IVRs.
>>
>> Any strong comments pro or against?
>>
>> What is your preferred virtualization system for such deployments?
>>
I host ~120 asterisk virtual serves on OpneVZ for more than 3 years,
runs great.  These are business customers so they can be very
demanding, allot of custom apps and such.  I host up to 150 phones in
virtual environment, if more is needed, I switch over to a physical
server.  This is just a personal preference, not limited virtually,
but customers with that many or more phones usually want a dedicated
server and can certainly afford the up charge.

Ensure your resources for each VM are enough for the voice apps you
want to run, be cautious about disk I/O like high volume recording,
database dips per call and such.  I just converted my physical
sip-routers to virtual about a week now on OpenVZ, so far so good.

Transcoding required higher proc and mem loads so partition resources
accordingly.  I do run Asterisk 10.0 conf bridge server virtual, SIP
only, no need for timing interface like DAHDI.  This is running well,
no problem that I'm aware of, just works.

I use VirtualBox and VMware for demo systems and lab work but OpenVZ
for production due to higher density you can achive per physical
server.

Whatever virtualization you want to use or are comfortable with,
ensure you know it backwards and forwards and what to do if it breaks
and how to increase resource when needed and by all means do your lab
work ahead of time.  You will not be able to simulate every situation
in the lab, but you can certainly stress test your environment with
high call volume, disk I/O recordings, databases operations, live
migration, upgrades, downgrades, backups, resource monitoring, alarm
triggers, test, test, test and then test some more.

Virtualizing voice systems has been around a long time, most virt
platforms can be implemented to perform well for voice, just got to do
your homework.

Good luck!.

JR
-- 
JR Richardson
Engineering for the Masses


From ibc at aliax.net  Tue Aug 28 19:12:01 2012
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Tue, 28 Aug 2012 19:12:01 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503C62E7.6010505@gmail.com>
References: <503C62E7.6010505@gmail.com>
Message-ID: <CALiegfnG6hCanmez5QFQ9Lnt702BdYQvwsS6tN7oZmGf-xvJ1A@mail.gmail.com>

2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com>:
> Hello,
>
> just asking to see your experience deploying sip platforms on virtual
> systems. So far I was running Kamailio in virtual machines and no problems,
> but I insisted that media servers to be on physical machines. Lately is more
> pressure from the market to go everything virtual.
>
> So the question is more about having everything on virtual systems, proxy
> and media server, where the media server can deal with transcoding,
> conference rooms and IVRs.
>
> Any strong comments pro or against?
>
> What is your preferred virtualization system for such deployments?

My company uses XEN virtual servers for various Kamailio and Rtp-Proxy
deployments with high traffic. All goes really ok if the host has
enough CPU and RAM, no latency problems at all.

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From miconda at gmail.com  Tue Aug 28 21:56:12 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Aug 2012 21:56:12 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120828163428.GC75367@putsch.kolbu.ws>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
Message-ID: <503D225C.4000304@gmail.com>

Hello,

On 8/28/12 6:34 PM, ?yvind Kolbu wrote:
> On 2012-08-24 at 10:48, ?yvind Kolbu wrote:
>> First impression is that everything is working now, thanks! At least
>> registrations are getting populated in both databases from both servers. As
>> I'm out traveling I havn't yet tested calling and whether or not the
>> failover works as expected. I assume setting usrloc's db_mode to 3 is
>> recommended in this kind of setup?
>>
>> For now I'm using:
>>      modparam("db_cluster", "cluster","data2=>voip1_data2=9s9p;voip2_data2=8s9p")
>> on my voip1 machine, and the line below on the voip2 machine:
>>      modparam("db_cluster", "cluster","data2=>voip2_data2=9s9p;voip1_data2=8s9p")
>>
>> The same configuration created coredumps on both machines previously.
> Have played with usrloc and db_cluster a bit in the lab and think I found
> two bugs. I have not inspected the actual sql-commands, but this is based
> in my observations. Both sql-servers are running postgresql 9.1.4.
>
> 1) If the second server lacks an entry in the location table it will never
>     be populated, while the primary is updated. This is a major flaw if
>     the second server has been unavailable and then is back again it will
>     never contain the same information as the primary.
>
>     I think this is because usrloc issues an UPDATE for the row it read,
>     but the second server can not UPDATE an non existent row.
>
>     Deleting the entry from the primary server will issue an INSERT to both
>     databases and then they are identical again, for that row at least.
>
> 2) If the primary lacks an entry and the second already has it, it will
>     get an extra entry for that user. After a while though the stale entry
>     will be pruned.
>
>     Probably similar reason as above. usrloc finds no entries in the table
>     when reading, thus issuing INSERT.
>
> Tried setting
>
>      modparam("usrloc", "db_check_update", 1)
>
> but to no avail.

what do you mean by the last line? This should be the solution to for 
the issue.

>
> Even enabling
>
>     modparam("usrloc", "db_update_as_insert", 1)
>
> will not fix issue #1.

This parameter is for some special backends, like cassadra, that don't 
take commands similar to sql update.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From oyvind.kolbu at usit.uio.no  Tue Aug 28 22:23:33 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Tue, 28 Aug 2012 22:23:33 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <503D225C.4000304@gmail.com>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
Message-ID: <20120828202333.GD75367@putsch.kolbu.ws>

On 2012-08-28 at 21:56, Daniel-Constantin Mierla wrote:
> On 8/28/12 6:34 PM, ?yvind Kolbu wrote:
> >
> > Tried setting
> >
> >      modparam("usrloc", "db_check_update", 1)
> >
> > but to no avail.
> 
> what do you mean by the last line? This should be the solution to for 
> the issue.

That I tried that option, but without success. Any other suggestions? I can
probably grab the sql statements received by the sql server.

> > Even enabling
> >
> >     modparam("usrloc", "db_update_as_insert", 1)
> >
> > will not fix issue #1.
> 
> This parameter is for some special backends, like cassadra, that don't 
> take commands similar to sql update.

OK.

-- 
?yvind Kolbu


From tayeb at vobradio.org  Tue Aug 28 23:17:20 2012
From: tayeb at vobradio.org (Tayeb Meftah)
Date: Wed, 29 Aug 2012 01:17:20 +0400
Subject: [SR-Users] [OT] OverSIP SIP Proxy released
In-Reply-To: <CALiegfnvMero6t7qTeLGWCC48wAAFdGB8Zf=Kt+SUiH1rOzaBw@mail.gmail.com>
References: <CALiegfnvMero6t7qTeLGWCC48wAAFdGB8Zf=Kt+SUiH1rOzaBw@mail.gmail.com>
Message-ID: <7626164428232717368@unknownmsgid>

Good jobs

Envoy? de mon iPhone

Le 28 ao?t 2012 ? 20:47, "I?aki Baz Castillo" <ibc at aliax.net> a ?crit :

> Hi all,
>
> I announce the first stable release of OverSIP SIP Proxy:
>
>  http://www.oversip.net
>
>
> OverSIP features:
> -----------------------------------------------------------------------------
>
> - SIP over UDP, TCP, TLS and WebSocket transports.
>
> - Full support for IPv4, IPv6 and DNS resolution (NAPTR, SRV, A, AAAA).
>
> - Fully programmable using Ruby programming language.
>
> - The perfect Outbound (RFC 5626) Edge Proxy.
>
> - Fully asynchronous event-based design.
>
>
> OverSIP is not a replacement for well-known proxies as Kamailio
> (sip-router). OverSIP does not implement basic proxy features such as
> Digest authentication, registration or parallel forking.
>
> OverSIP can behave as an Outbound Edge Proxy between clients and SIP
> registrars, or can be placed anywhere for building custom SIP /
> WebSocket applications.
>
>
> Cheers.
>
>
> --
> I?aki Baz Castillo
> <ibc at aliax.net>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


From miconda at gmail.com  Wed Aug 29 07:38:20 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 29 Aug 2012 07:38:20 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120828202333.GD75367@putsch.kolbu.ws>
References: <20120822133240.GF16374@putsch.kolbu.ws>
	<20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
Message-ID: <503DAACC.5080808@gmail.com>


On 8/28/12 10:23 PM, ?yvind Kolbu wrote:
> On 2012-08-28 at 21:56, Daniel-Constantin Mierla wrote:
>> On 8/28/12 6:34 PM, ?yvind Kolbu wrote:
>>> Tried setting
>>>
>>>       modparam("usrloc", "db_check_update", 1)
>>>
>>> but to no avail.
>> what do you mean by the last line? This should be the solution to for
>> the issue.
> That I tried that option, but without success. Any other suggestions? I can
> probably grab the sql statements received by the sql server.

seeing the sql commands would help, also the log messages with debug=3 
for that particular case.

Cheers,
Daniel

>
>>> Even enabling
>>>
>>>      modparam("usrloc", "db_update_as_insert", 1)
>>>
>>> will not fix issue #1.
>> This parameter is for some special backends, like cassadra, that don't
>> take commands similar to sql update.
> OK.
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Wed Aug 29 07:46:28 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 29 Aug 2012 07:46:28 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <CAE=KcrjJJT-esq65aoTTNJApPxTv9EPrFt9dzzzd9uUy_bpeFA@mail.gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
	<503CEA26.4000606@gmail.com>
	<CAE=KcrjJJT-esq65aoTTNJApPxTv9EPrFt9dzzzd9uUy_bpeFA@mail.gmail.com>
Message-ID: <503DACB4.7060502@gmail.com>


On 8/28/12 6:09 PM, Jason Penton wrote:
>
> No. We use solaris11.
>

IIRC, solaris 11 was free for evaluation purposes, has that changed?

BTW, since mainstream opensolaris was discontinued, anyone knows what is 
the best derivative (if that is at all)?

Cheers,
Daniel

> But yes on any hardware
>
> On Aug 28, 2012 5:56 PM, "Daniel-Constantin Mierla" <miconda at gmail.com 
> <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     On 8/28/12 10:33 AM, Jason Penton wrote:
>>     Hey Daniel,
>>
>>     We use Solaris virtualisation and it works great. The zones (VMs
>>     per se) are lightweight, easy to administer and rock solid.
>>
>>     btw, common misconceptions are that you need sun (oracle)
>>     hardware and that the os is not free. These are both false.
>
>     so you use opensolaris, I guess, and then it can be any intel/amd
>     arch server (e.g., dell, hp)?
>
>     Cheers,
>     Daniel
>
>>
>>     cheers
>>     Jason
>>
>>     On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock
>>     <carsten at ng-voice.com <mailto:carsten at ng-voice.com>> wrote:
>>
>>         Hi Daniel,
>>
>>         here's from my personal experience:
>>         Our setup at ng-voice is a little weird sometimes: We've
>>         rented some
>>         virtual servers at a german provider (who uses Xen). On these
>>         virtual
>>         servers we've installed OpenVz, which for us is absolutely
>>         great, if
>>         you are just working with Linux-Servers. While Xen is a rather
>>         complete virtualization, OpenVz is lightweight and comes in
>>         handy, if
>>         you just want to logically separate servers. We've got each IMS
>>         component (P-/I-/S-CSCF, HSS, Application-Servers, Databases)
>>         running
>>         on a dedicated OpenVz Container, which is really great. We've
>>         even got
>>         a CentOs-Container running on a Debian OpenVz, which is started
>>         "on-demand" in order to build RPM-Packages. With OpenVz you
>>         can even
>>         move Containers from one host to another, theoretically with zero
>>         downtime (doesn't work with SEMS, don't know about other
>>         software).
>>         For our IMS-setup, we work with RTP-Relaying, which works
>>         great within
>>         virtualization, i cannot complain.
>>
>>         At another customer (a fibre provider in Germany), we're
>>         running all
>>         the infrastructure on Xen-only. An infrastructure provider
>>         takes care
>>         of the administration, but those servers run poorly
>>         (RTP-Relaying is
>>         okay but everything else is really slow).
>>
>>         Conclusion for me: VoIP on virtual servers can work great,
>>         but the
>>         virtualization infrastructure needs to be administered
>>         properly which
>>         may not be an easy task, if you are new in this subject.
>>
>>         Kind regards,
>>         Carsten
>>
>>         2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com
>>         <mailto:miconda at gmail.com>>:
>>         > Hello,
>>         >
>>         > just asking to see your experience deploying sip platforms
>>         on virtual
>>         > systems. So far I was running Kamailio in virtual machines
>>         and no problems,
>>         > but I insisted that media servers to be on physical
>>         machines. Lately is more
>>         > pressure from the market to go everything virtual.
>>         >
>>         > So the question is more about having everything on virtual
>>         systems, proxy
>>         > and media server, where the media server can deal with
>>         transcoding,
>>         > conference rooms and IVRs.
>>         >
>>         > Any strong comments pro or against?
>>         >
>>         > What is your preferred virtualization system for such
>>         deployments?
>>         >
>>         > Cheers,
>>         > Daniel
>>         >
>>         > --
>>         > Daniel-Constantin Mierla - http://www.asipto.com
>>         > http://twitter.com/#!/miconda
>>         <http://twitter.com/#%21/miconda> -
>>         http://www.linkedin.com/in/miconda
>>         > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>         http://asipto.com/u/kat
>>         >
>>         >
>>         > _______________________________________________
>>         > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>         mailing list
>>         > sr-users at lists.sip-router.org
>>         <mailto:sr-users at lists.sip-router.org>
>>         > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>         --
>>         Carsten Bock
>>         CEO (Gesch?ftsf?hrer)
>>
>>         ng-voice GmbH
>>         Schomburgstr. 80
>>         D-22767 Hamburg / Germany
>>
>>         http://www.ng-voice.com
>>         mailto:carsten at ng-voice.com <mailto:carsten at ng-voice.com>
>>
>>         Office +49 40 34927219 <tel:%2B49%2040%2034927219>
>>         Fax +49 40 34927220 <tel:%2B49%2040%2034927220>
>>
>>         Sitz der Gesellschaft: Hamburg
>>         Registergericht: Amtsgericht Hamburg, HRB 120189
>>         Gesch?ftsf?hrer: Carsten Bock
>>         Ust-ID: DE279344284
>>
>>         Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>>         http://www.ng-voice.com/imprint/
>>
>>         _______________________________________________
>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>         mailing list
>>         sr-users at lists.sip-router.org
>>         <mailto:sr-users at lists.sip-router.org>
>>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>     This email is subject to the disclaimer of Smile Communications (PTY) Ltd. athttp://www.smilecoms.com/disclaimer
>>
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla -http://www.asipto.com
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>
> This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer
>
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From jason.penton at smilecoms.com  Wed Aug 29 08:49:35 2012
From: jason.penton at smilecoms.com (Jason Penton)
Date: Wed, 29 Aug 2012 08:49:35 +0200
Subject: [SR-Users] [ot] virtualization systems
In-Reply-To: <503DACB4.7060502@gmail.com>
References: <503C62E7.6010505@gmail.com>
	<CAOCjumG4ei-PK_6o_fCJ=HuH_F2nMaXOPE=yBmNCW4tuncRk6Q@mail.gmail.com>
	<CAE=Kcriaf1bkZ0aGJy5dhdwNLdd=arB4mWnOX0ty=pQEv6LfCQ@mail.gmail.com>
	<503CEA26.4000606@gmail.com>
	<CAE=KcrjJJT-esq65aoTTNJApPxTv9EPrFt9dzzzd9uUy_bpeFA@mail.gmail.com>
	<503DACB4.7060502@gmail.com>
Message-ID: <CAE=Kcri3EZ1QqTGdpMLGN4iDUiH-UWZkM9LU0=mjooi+FAmC=A@mail.gmail.com>

Hi Daniel,

not evaluation, but if you use it in production then you are required to
buy a OS support contract. We however have standardised on Sun hardware
which includes support for Solaris 11 in our production environment. So,
yes a good point you make if you are going to be using in production on
non-sun hardware you will need to pay Oracle ;(

cheers
Jason

On Wed, Aug 29, 2012 at 7:46 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>
> On 8/28/12 6:09 PM, Jason Penton wrote:
>
> No. We use solaris11.
>
>
> IIRC, solaris 11 was free for evaluation purposes, has that changed?
>
> BTW, since mainstream opensolaris was discontinued, anyone knows what is
> the best derivative (if that is at all)?
>
> Cheers,
> Daniel
>
>
>  But yes on any hardware
> On Aug 28, 2012 5:56 PM, "Daniel-Constantin Mierla" <miconda at gmail.com>
> wrote:
>
>>  Hello,
>>
>> On 8/28/12 10:33 AM, Jason Penton wrote:
>>
>> Hey Daniel,
>>
>> We use Solaris virtualisation and it works great. The zones (VMs per se)
>> are lightweight, easy to administer and rock solid.
>>
>> btw, common misconceptions are that you need sun (oracle) hardware and
>> that the os is not free. These are both false.
>>
>>
>> so you use opensolaris, I guess, and then it can be any intel/amd arch
>> server (e.g., dell, hp)?
>>
>> Cheers,
>> Daniel
>>
>>
>> cheers
>> Jason
>>
>> On Tue, Aug 28, 2012 at 9:59 AM, Carsten Bock <carsten at ng-voice.com>wrote:
>>
>>> Hi Daniel,
>>>
>>> here's from my personal experience:
>>> Our setup at ng-voice is a little weird sometimes: We've rented some
>>> virtual servers at a german provider (who uses Xen). On these virtual
>>> servers we've installed OpenVz, which for us is absolutely great, if
>>> you are just working with Linux-Servers. While Xen is a rather
>>> complete virtualization, OpenVz is lightweight and comes in handy, if
>>> you just want to logically separate servers. We've got each IMS
>>> component (P-/I-/S-CSCF, HSS, Application-Servers, Databases) running
>>> on a dedicated OpenVz Container, which is really great. We've even got
>>> a CentOs-Container running on a Debian OpenVz, which is started
>>> "on-demand" in order to build RPM-Packages. With OpenVz you can even
>>> move Containers from one host to another, theoretically with zero
>>> downtime (doesn't work with SEMS, don't know about other software).
>>> For our IMS-setup, we work with RTP-Relaying, which works great within
>>> virtualization, i cannot complain.
>>>
>>> At another customer (a fibre provider in Germany), we're running all
>>> the infrastructure on Xen-only. An infrastructure provider takes care
>>> of the administration, but those servers run poorly (RTP-Relaying is
>>> okay but everything else is really slow).
>>>
>>> Conclusion for me: VoIP on virtual servers can work great, but the
>>> virtualization infrastructure needs to be administered properly which
>>> may not be an easy task, if you are new in this subject.
>>>
>>> Kind regards,
>>> Carsten
>>>
>>> 2012/8/28 Daniel-Constantin Mierla <miconda at gmail.com>:
>>>  > Hello,
>>> >
>>> > just asking to see your experience deploying sip platforms on virtual
>>> > systems. So far I was running Kamailio in virtual machines and no
>>> problems,
>>> > but I insisted that media servers to be on physical machines. Lately
>>> is more
>>> > pressure from the market to go everything virtual.
>>> >
>>> > So the question is more about having everything on virtual systems,
>>> proxy
>>> > and media server, where the media server can deal with transcoding,
>>> > conference rooms and IVRs.
>>> >
>>> > Any strong comments pro or against?
>>> >
>>> > What is your preferred virtualization system for such deployments?
>>> >
>>> > Cheers,
>>> > Daniel
>>> >
>>> > --
>>> > Daniel-Constantin Mierla - http://www.asipto.com
>>> > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>> http://asipto.com/u/kat
>>> >
>>> >
>>> > _______________________________________________
>>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> > sr-users at lists.sip-router.org
>>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
>>>  --
>>> Carsten Bock
>>> CEO (Gesch?ftsf?hrer)
>>>
>>> ng-voice GmbH
>>> Schomburgstr. 80
>>> D-22767 Hamburg / Germany
>>>
>>> http://www.ng-voice.com
>>> mailto:carsten at ng-voice.com
>>>
>>> Office +49 40 34927219 <%2B49%2040%2034927219>
>>> Fax +49 40 34927220 <%2B49%2040%2034927220>
>>>
>>> Sitz der Gesellschaft: Hamburg
>>> Registergericht: Amtsgericht Hamburg, HRB 120189
>>> Gesch?ftsf?hrer: Carsten Bock
>>> Ust-ID: DE279344284
>>>
>>> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
>>> http://www.ng-voice.com/imprint/
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>> This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer
>>
>>
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>
>>   This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer
>
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>

This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer

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From serusers at iptel.org  Wed Aug 29 11:50:31 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Wed, 29 Aug 2012 03:50:31 -0600
Subject: [SR-Users] Work offer inside
Message-ID: <1449140021.0SASU3EC849089@rfwhawnrfpnlvti.egdewnnf.org>

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Content preview:  I would like to take this time to welcome you to our hiring
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From oyvind.kolbu at usit.uio.no  Wed Aug 29 13:23:57 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Wed, 29 Aug 2012 13:23:57 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <503DAACC.5080808@gmail.com>
References: <20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
Message-ID: <20120829112357.GE75367@putsch.kolbu.ws>

On 2012-08-29 at 07:38, Daniel-Constantin Mierla wrote:
> seeing the sql commands would help, also the log messages with debug=3 

Now with "db_check_update" enabled. I've deleted the row for username 2549619
in the voip_location table from the secondary server between then re-registration.

I think the problem is that the check is done using only one of the "read table"
connections, and not done for each of the "write table" connections.

Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:436]: use table (voip_location) - cluster [data2]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/1)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (8/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (7/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:299]: executing db cluster query command
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:300]: serial operation - cluster [data2] (8/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 7 chars, out: 7 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7c9dfd0 (7) - [select contact,expires,q,callid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified,ruid,instance,reg_id from voip_location where username='2549619' AND domain='hometest.voip.uio.no' or
der by q]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0xb7cd2d90
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:494]: 0xb7c9dfd0 PQresultStatus(PGRES_TUPLES_OK) PQgetResult(0x9058e78)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:108]: 16 columns returned from the query
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:155]: allocate 64 bytes for result names at 0xb7cd2db8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:165]: allocate 64 bytes for result types at 0xb7cd2e00
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[0] at 0xb7ca14d0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca14d0)[0]=[contact]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[1] at 0xb7ca16e8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca16e8)[1]=[expires]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:158]: use DB1_DATETIME result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[2] at 0xb7ca15d0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca15d0)[2]=[q]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:151]: use DB1_DOUBLE result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[3] at 0xb7ca1530
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca1530)[3]=[callid]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[4] at 0xb7ca14a0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca14a0)[4]=[cseq]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[5] at 0xb7ca14e0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca14e0)[5]=[flags]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[6] at 0xb7ca16a8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca16a8)[6]=[cflags]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[7] at 0xb7ca14b0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca14b0)[7]=[user_agent]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[8] at 0xb7ca14c0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca14c0)[8]=[received]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[9] at 0xb7ca14f0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7ca14f0)[9]=[path]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[10] at 0xb7cd3098
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7cd3098)[10]=[socket]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[11] at 0xb7cd30a8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7cd30a8)[11]=[methods]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[12] at 0xb7cd30b8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7cd30b8)[12]=[last_modified]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:158]: use DB1_DATETIME result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[13] at 0xb7cd30c8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7cd30c8)[13]=[ruid]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[14] at 0xb7cd30d8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7cd30d8)[14]=[instance]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[15] at 0xb7cd30e8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0xb7cd30e8)[15]=[reg_id]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:222]: allocate for 16 columns 64 bytes in row buffer at 0xb7cd30f8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:184]: allocate 8 bytes for rows at 0xb7cd3140
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,0)=[sip:2549619 at 193.157.149.56]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][0] Column[contact]=[sip:2549619 at 193.157.149.56]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,1)=[2012-08-29 13:20:11]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][1] Column[expires]=[2012-08-29 13:20:11]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,2)=[-1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][2] Column[q]=[-1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,3)=[94f2f534-e3cc629f-5bd8343e at 193.157.149.56]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][3] Column[callid]=[94f2f534-e3cc629f-5bd8343e at 193.157.149.56]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,4)=[14]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][4] Column[cseq]=[14]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,5)=[0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][5] Column[flags]=[0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,6)=[0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][6] Column[cflags]=[0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,7)=[PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][7] Column[user_agent]=[PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,8)=[]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,9)=[]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,10)=[udp:129.240.253.197:5060]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][10] Column[socket]=[udp:129.240.253.197:5060]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,11)=[8159]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][11] Column[methods]=[8159]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,12)=[2012-08-29 13:18:11]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][12] Column[last_modified]=[2012-08-29 13:18:11]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,13)=[uloc-503dfa6f-7959-1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][13] Column[ruid]=[uloc-503dfa6f-7959-1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,14)=[]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:243]: PQgetvalue(0xb7c9dfd0,0,15)=[0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:252]: [0][15] Column[reg_id]=[0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:119]: allocate 320 bytes for row values at 0xb7cd3150
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:2549619 at 193.157.149.56]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:127]: allocate 27 bytes memory for STRING at 0xb7cd32b0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:157]: converting DATETIME [2012-08-29 13:20:11]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:106]: converting DOUBLE [-1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:117]: converting STRING [94f2f534-e3cc629f-5bd8343e at 193.157.149.56]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:127]: allocate 42 bytes memory for STRING at 0xb7cd32d8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:73]: converting INT [14]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:73]: converting INT [0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:73]: converting INT [0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:117]: converting STRING [PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:127]: allocate 43 bytes memory for STRING at 0xb7cd3310
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:56]: converting NULL value
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:56]: converting NULL value
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:117]: converting STRING [udp:129.240.253.197:5060]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:127]: allocate 25 bytes memory for STRING at 0xb7cd3348
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:73]: converting INT [8159]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:157]: converting DATETIME [2012-08-29 13:18:11]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:117]: converting STRING [uloc-503dfa6f-7959-1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:127]: allocate 21 bytes memory for STRING at 0xb7cd3370
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:56]: converting NULL value
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_val.c:73]: converting INT [0]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_res.c:267]: freeing row buffer at 0xb7cd30f8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x9058e78) result set
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [socket_info.c:583]: grep_sock_info - checking if host==us: 15==9 && [129.240.253.197] == [127.0.0.1]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise 0) matches port 5060
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [socket_info.c:583]: grep_sock_info - checking if host==us: 15==15 && [129.240.253.197] == [129.240.253.197]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise 0) matches port 5060
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:284]: executing db cluster free-result command
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:81]: freeing 16 columns
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0xb7ca14d0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0xb7ca16e8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0xb7ca15d0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0xb7ca1530
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0xb7ca14a0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[5] at 0xb7ca14e0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[6] at 0xb7ca16a8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[7] at 0xb7ca14b0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[8] at 0xb7ca14c0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[9] at 0xb7ca14f0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[10] at 0xb7cd3098
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[11] at 0xb7cd30a8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[12] at 0xb7cd30b8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[13] at 0xb7cd30c8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[14] at 0xb7cd30d8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[15] at 0xb7cd30e8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:94]: freeing result names at 0xb7cd2db8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:99]: freeing result types at 0xb7cd2e00
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:54]: freeing 1 rows
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:68]: free VAL_STRING[0] 'sip:2549619 at 193.157.149.56' at 0xb7cd32b0
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:68]: free VAL_STRING[3] '94f2f534-e3cc629f-5bd8343e at 193.157.149.56' at 0xb7cd32d8
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:68]: free VAL_STRING[7] 'PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054' at 0xb7cd3310
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:68]: free VAL_STRING[10] 'udp:129.240.253.197:5060' at 0xb7cd3348
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:68]: free VAL_STRING[13] 'uloc-503dfa6f-7959-1' at 0xb7cd3370
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_row.c:97]: freeing row values at 0xb7cd3150
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:62]: freeing rows at 0xb7cd3140
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:136]: freeing result set at 0xb7cd2d90
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: registrar [save.c:564]: 1 valid contacts
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: registrar [save.c:586]: 1 contacts after commit
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [sruid.c:176]: new sruid is [uloc-503dfa6f-795c-1] (1 / 20)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:436]: use table (voip_location) - cluster [data2]
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/1)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (8/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (7/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:356]: executing db cluster update command
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:357]: parallel operation - cluster [data2] (9/0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 42 chars, out: 42 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 24 chars, out: 24 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 7 chars, out: 7 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 26 chars, out: 26 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 41 chars, out: 41 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7c9df08 (7) - [update voip_location set expires='2012-08-29 13:21:11',q=-1.00     ,cseq=15,flags=0,cflags=0,user_agent='PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054',received=NULL,path=NULL,socket='udp:129.240.253.197:5060',methods=8159,last_modified='2012-08-29 13:19:11',ruid='uloc-503dfa6f-7959-1',instance=NULL,reg_id=0 where username='2549619' AND contact='sip:2549619 at 193.157.149.56' AND callid='94f2f534-e3cc629f-5bd8343e at 193.157.149.56' AND domain='hometest.voip.uio.no']
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0xb7cd2d90
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:494]: 0xb7c9df08 PQresultStatus(PGRES_COMMAND_OK) PQgetResult(0x9058e78)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x9058e78) result set
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:81]: freeing 0 columns
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:136]: freeing result set at 0xb7cd2d90
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:357]: parallel operation - cluster [data2] (9/1)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 42 chars, out: 42 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 24 chars, out: 24 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 7 chars, out: 7 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 26 chars, out: 26 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 41 chars, out: 41 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7c9dfd0 (7) - [update voip_location set expires='2012-08-29 13:21:11',q=-1.00     ,cseq=15,flags=0,cflags=0,user_agent='PolycomSoundPointIP-SPIP_331-UA/3.2.1.0054',received=NULL,path=NULL,socket='udp:129.240.253.197:5060',methods=8159,last_modified='2012-08-29 13:19:11',ruid='uloc-503dfa6f-7959-1',instance=NULL,reg_id=0 where username='2549619' AND contact='sip:2549619 at 193.157.149.56' AND callid='94f2f534-e3cc629f-5bd8343e at 193.157.149.56' AND domain='hometest.voip.uio.no']
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0xb7cd2d90
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:494]: 0xb7c9dfd0 PQresultStatus(PGRES_COMMAND_OK) PQgetResult(0x9058e78)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x9058e78) result set
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:81]: freeing 0 columns
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [db_res.c:136]: freeing result set at 0xb7cd2d90
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: db_cluster [dbcl_api.c:391]: executing db cluster affected-rows command
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: registrar [reply.c:368]: created Contact HF: Contact: <sip:2549619 at 193.157.149.56>;expires=120

Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: sl [sl.c:278]: reply in stateless mode (sl)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [msg_translator.c:206]: check_via_address(193.157.149.56, 193.157.149.56, 0)
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list 0xb5d23220
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil)
Aug 29 13:19:11 voipt1-edge1 last message repeated 4 times
Aug 29 13:19:11 voipt1-edge1 /nett/kamailio/current/sbin/kamailio[31068]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil)


-- 
?yvind Kolbu


From agranig at sipwise.com  Wed Aug 29 14:14:51 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 29 Aug 2012 14:14:51 +0200
Subject: [SR-Users] [OT] sip:provider CE v2.6 is out!
Message-ID: <503E07BB.2040902@sipwise.com>

Hi all,

I'm excited to announce the release of the Sipwise sip:provider
Community Edition v2.6, a free and open source turn-key platform, which
uses Kamailio, Sems and Asterisk in its core to provide a full-blown and
feature-rich VoIP soft-switch.

http://www.sipwise.com/news/announcements/spce-v2_6-release/

New core features are *full presence/xcap/im support* together with the
awesome Jitsi client, allowing you to deploy your own Skype-like service
for encryted voice/video communication, buddy lists, instant messaging,
screen sharing and remote desktop control. We also support *media
features for error announcements*, like busy, blocked, unknown number etc.

Check our VM images to get started within a couple of minutes and learn
what can be built with the power of Kamailio as a stateless
load-balancer and a stateful proxy/registrar, and Sems as an SBC and
application server.

http://www.sipwise.com/products/spce/quickinstall/

Thanks a lot to all developers, contributors and members of this mailing
list to make this possible and for helping us out!

Andreas

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From fchahrour at vanrise.com  Wed Aug 29 15:31:05 2012
From: fchahrour at vanrise.com (Fatima Chahrour~Vanrise Support)
Date: Wed, 29 Aug 2012 16:31:05 +0300
Subject: [SR-Users] FW: Kamailio as sipRedirect
Message-ID: <04ee01cd85ea$8af4e880$a0deb980$@com>

Dears,

 

I have an installed 3CX where I want to generate a call from it to the
Kamailo to get a list of suppliers to terminate the call to.

Thus, the Kamailio should act as a sip redirect to the 3CX (for routing
decision).

 

Have you tried such configuration before? And how do I configure the 3CX on
the Kamailio?

 

Regards,

F Charour

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From anca.vamanu at 1and1.ro  Wed Aug 29 18:42:27 2012
From: anca.vamanu at 1and1.ro (Anca Vamanu)
Date: Wed, 29 Aug 2012 19:42:27 +0300
Subject: [SR-Users] Kamailio Crash using Cassandra 1.1.2 during	SIP
	unregister
In-Reply-To: <D8261FE064C78B44B77EF7C4569D6CBF0D11DDC3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
References: <D8261FE064C78B44B77EF7C4569D6CBF0D11D150@DB3PRD0510MB380.eurprd05.prod.outlook.com>
	<D8261FE064C78B44B77EF7C4569D6CBF0D11DDC3@DB3PRD0510MB380.eurprd05.prod.outlook.com>
Message-ID: <503E4673.9010108@1and1.ro>

Hi Bou,

You are right, those columns were added this year and the schema was not 
updated in the cassandra file also. I have committed the fix for that file.
I will investigate also why it crashed in this case, as it shouldn't 
crash in any case.

Thanks and regards,
Anca Vamanu


On 08/24/2012 09:07 PM, Boudewyn Ligthart wrote:
>
> An update on this issue:
>
> The schema that is used by the cassandra module seems to be missing a 
> few fields, the correct schema for location should be:
>
> *************************
>
> /callid(string) cflags(int) contact(string) cseq(int) 
> expires(timestamp) flags(int) last_modified(int) methods(int) 
> path(string) q(double) received(string) socket(string) 
> user_agent(string) username(string) ruid(string) instance(string)/
>
> /username/
>
> /contact/
>
> *************************
>
> //
>
> The two fields : ruid and instance where missing, please append the 
> documentation. (utils/kamctl/dbcassandra/openser/location)
>
> Thanks,
>
> Bou
>
> **
>
> *
>
> *
>
> *From:*sr-users-bounces at lists.sip-router.org 
> [mailto:sr-users-bounces at lists.sip-router.org] *On Behalf Of *Boudewyn 
> Ligthart
> *Sent:* 22 August 2012 16:34
> *To:* sr-users at lists.sip-router.org
> *Subject:* [SR-Users] Kamailio Crash using Cassandra 1.1.2 during SIP 
> unregister
>
> Hi,
>
> Kamailio crash using Cassandra 1.1.2 during SIP unregister 
> (registering works fine)
>
> The version of Kamailio is the latest GIT version:
>
> /version: kamailio 3.4.0-dev3 (x86_64/linux) 10327c/
>
> /flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, 
> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 
> SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, 
> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, 
> HAVE_RESOLV_RES/
>
> /ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB/
>
> /poll method support: poll, epoll_lt, epoll_et, sigio_rt, select./
>
> /id: 10327c/
>
> /compiled on 14:53:11 Aug 22 2012 with gcc 4.6.3/
>
> BT of Core :
>
> /Using host libthread_db library 
> "/lib/x86_64-linux-gnu/libthread_db.so.1"./
>
> /Core was generated by `kamailio -ddddd'./
>
> /Program terminated with signal 11, Segmentation fault./
>
> /#0 fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at 
> mem/f_malloc.c:527/
>
> /527 if(unlikely(f->u.nxt_free!=NULL)) {/
>
> /(gdb) bt/
>
> /#0 fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at 
> mem/f_malloc.c:527/
>
> /#1 0x00007fa55d35ac53 in db_free_columns (_r=0x7fa55db38c30) at 
> db_res.c:86/
>
> /#2 0x00007fa55d83f87d in db_cassa_query (_h=<optimized out>, 
> _k=0x7fa55adc6aac, _op=<optimized out>, _v=<optimized out>, 
> _c=0x7fff72ef0e30, _n=<optimized out>,/
>
> /_nc=16, _o=0x7fa55afca990, _r=0x7fff72ef0f10) at dbcassa_base.cpp:659/
>
> /#3 0x00007fa55adb58eb in db_load_urecord (_c=0x7fa55db218d0, 
> _d=0x7fa5576eee38, _aor=0x7fff72ef11b0) at udomain.c:592/
>
> /#4 0x00007fa55adb93cd in get_urecord (_d=<optimized out>, 
> _aor=0x7fff72ef11b0, _r=0x7fff72ef1200) at udomain.c:1017/
>
> /#5 0x00007fa55ab91698 in lookup (_m=0x7fa55db40c48, 
> _d=0x7fa5576eee38, _uri=<optimized out>) at lookup.c:156/
>
> /#6 0x00007fa55ab957f2 in w_lookup (_m=0x7fa55db40c48, 
> _d=0x7fa5576eee38 "\350\355nW\245\177", _uri=<optimized out>) at 
> reg_mod.c:437/
>
> /#7 0x000000000041c624 in do_action (h=0x7fff72ef17f0, 
> a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1153/
>
> /#8 0x000000000041b493 in run_actions (h=0x7fff72ef17f0, 
> a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1646/
>
> /#9 0x0000000000423a24 in run_actions_safe (h=<optimized out>, 
> a=<optimized out>, msg=<optimized out>) at action.c:1710/
>
> /#10 0x00000000004d79a0 in rval_get_int (h=0x7fff72ef2790, 
> msg=<optimized out>, i=0x7fff72ef1c30, rv=<optimized out>, 
> cache=<optimized out>) at rvalue.c:920/
>
> /#11 0x00000000004dc82c in rval_expr_eval_int (h=0x7fff72ef2790, 
> msg=0x7fa55db40c48, res=0x7fff72ef1c30, rve=0x7fa55db23218) at 
> rvalue.c:1914/
>
> /#12 0x00000000004dc8f9 in rval_expr_eval_int (h=0x7fff72ef2790, 
> msg=0x7fa55db40c48, res=0x7fff72ef20d0, rve=0x7fa55db238e8) at 
> rvalue.c:1922/
>
> /#13 0x000000000041cb66 in do_action (h=0x7fff72ef2790, 
> a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1123/
>
> /#14 0x000000000041b493 in run_actions (h=0x7fff72ef2790, 
> a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1646/
>
> /#15 0x000000000041c7e1 in do_action (h=0x7fff72ef2790, 
> a=0x7fa55db15b38, msg=0x7fa55db40c48) at action.c:763/
>
> /#16 0x000000000041b493 in run_actions (h=0x7fff72ef2790, 
> a=0x7fa55db118c8, msg=0x7fa55db40c48) at action.c:1646/
>
> /#17 0x0000000000423ab0 in run_top_route (a=0x7fa55db118c8, 
> msg=0x7fa55db40c48, c=<optimized out>) at action.c:1731/
>
> /#18 0x00000000004b39ee in receive_msg (buf=<optimized out>, 
> len=<optimized out>, rcv_info=<optimized out>) at receive.c:209/
>
> /#19 0x000000000054de1a in udp_rcv_loop () at udp_server.c:544/
>
> /#20 0x0000000000478879 in main_loop () at main.c:1635/
>
> /#21 0x000000000041b002 in main (argc=<optimized out>, argv=<optimized 
> out>) at main.c:2548/
>
> Thanks,
>
> Bou
>


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From miconda at gmail.com  Wed Aug 29 20:16:35 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 29 Aug 2012 20:16:35 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120829112357.GE75367@putsch.kolbu.ws>
References: <20120822133743.GG16374@putsch.kolbu.ws> <5034E61C.60600@gmail.com>
	<20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
Message-ID: <503E5C83.2050102@gmail.com>

Hello,

On 8/29/12 1:23 PM, ?yvind Kolbu wrote:
> On 2012-08-29 at 07:38, Daniel-Constantin Mierla wrote:
>> seeing the sql commands would help, also the log messages with debug=3
> Now with "db_check_update" enabled. I've deleted the row for username 2549619
> in the voip_location table from the secondary server between then re-registration.
>
> I think the problem is that the check is done using only one of the "read table"
> connections, and not done for each of the "write table" connections.
is usrloc module configured in db only mode?

Read operation cannot be done in 'parallel' fashion of the cluster, it 
is only serial or round robin, stopping at first successful query (even 
when there is no data records in the result of the query).

If usrloc keeps also the records in memory, there is no select done at 
runtime, only updates, which are write operations.

db_check_update is only for SQL UPDATE command, in MySQL you can see the 
number of affected rows and if it is 0, then the module does an insert 
(another write operation).

So, the question is why do you think the check is done on one 'read 
table', or you don't refer to the check of affected rows for UPDATE?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Wed Aug 29 20:21:40 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 29 Aug 2012 20:21:40 +0200
Subject: [SR-Users] FW: Kamailio as sipRedirect
In-Reply-To: <04ee01cd85ea$8af4e880$a0deb980$@com>
References: <04ee01cd85ea$8af4e880$a0deb980$@com>
Message-ID: <503E5DB4.4040509@gmail.com>

Hello,

On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote:
>
> Dears,
>
> I have an installed 3CX where I want to generate a call from it to the 
> Kamailo to get a list of suppliers to terminate the call to.
>
> Thus, the Kamailio should act as a sip redirect to the 3CX (for 
> routing decision).
>
> Have you tried such configuration before? And how do I configure the 
> 3CX on the Kamailio?
>
I don't understand what you expect from 'configure the 3CX on the 
Kamailio', can you add more details ?

For sending a redirect, practically you have to update request uri and 
then send the 3xx reply, like:

$ru = "sip:newnumber at newdomain.com";
send_reply("302", "Redirected";

If you need to put more contacts in the redirect reply, then add:

km_append_branch("uri");

uri parameter can contain variables.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From Fchahrour at vanrise.com  Wed Aug 29 21:15:33 2012
From: Fchahrour at vanrise.com (Fatima Chahrour)
Date: Wed, 29 Aug 2012 22:15:33 +0300
Subject: [SR-Users] FW: Kamailio as sipRedirect
In-Reply-To: <503E5DB4.4040509@gmail.com>
References: <04ee01cd85ea$8af4e880$a0deb980$@com> <503E5DB4.4040509@gmail.com>
Message-ID: <3CB7D99A-D567-4FDF-A17B-B7718E554835@vanrise.com>

Hello Danial,

Hereby details:
My current working 1st scenario is a call generated from A-party x-lite registered on kamalio, this call is routed based on kamailio lcr routing module which add prefix and send prefix to 3CX then 3CX forward the call to B-party x-lite, registered on 3CX, successfully.

In kamailio.cfg routing logic I only used :

...
If (is_method("INVITE"))
{ if(!load_gateway("1")) {
sl_send_reply("503", "unable to load gateways");
Exit;  }

{ if(! next_gateway()) {
sl_send_reply("503", "unable to find gateways");
Exit;  }
 Route(1);
....

Now, my target is to register also the A-party x-lite to 3CX where I added a rule in 3CX to forward the call comming to kamailio to take the route desicion and act.

So, here I need to know how can I tell kamailio to accept the call from 3CX? this is what I meant by configuring 3CX on kamailio.

I have weakness in the routing logic configuration, could you please help in this too! What is the required replacement on the above to be done?

Moreover, I appreciate if you provide with docs or tutorials that can help me understand more kamailio's routing logic functions!

Much appreciated.

Thanks in advance.

F Chahrour

Sent from iPhone

On Aug 29, 2012, at 9:21 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hello,
> 
> On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote:
>> Dears,
>>  
>> I have an installed 3CX where I want to generate a call from it to the Kamailo to get a list of suppliers to terminate the call to.
>> Thus, the Kamailio should act as a sip redirect to the 3CX (for routing decision).
>>  
>> Have you tried such configuration before? And how do I configure the 3CX on the Kamailio?
>>  
> I don't understand what you expect from 'configure the 3CX on the Kamailio', can you add more details ?
> 
> For sending a redirect, practically you have to update request uri and then send the 3xx reply, like:
> 
> $ru = "sip:newnumber at newdomain.com";
> send_reply("302", "Redirected";
> 
> If you need to put more contacts in the redirect reply, then add:
> 
> km_append_branch("uri");
> 
> uri parameter can contain variables.
> 
> Cheers,
> Daniel
>> 
> 
> -- 
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From oyvind.kolbu at usit.uio.no  Wed Aug 29 21:15:45 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Wed, 29 Aug 2012 21:15:45 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <503E5C83.2050102@gmail.com>
References: <20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
Message-ID: <20120829191545.GJ75367@putsch.kolbu.ws>

On 2012-08-29 at 20:16, Daniel-Constantin Mierla wrote:
> is usrloc module configured in db only mode?

Yes, db_mode is 3.

> Read operation cannot be done in 'parallel' fashion of the cluster, it 
> is only serial or round robin, stopping at first successful query (even 
> when there is no data records in the result of the query).

Not related to this thread, but would it be possible to force it to try
until data is found?

> If usrloc keeps also the records in memory, there is no select done at 
> runtime, only updates, which are write operations.
> 
> db_check_update is only for SQL UPDATE command, in MySQL you can see the 
> number of affected rows and if it is 0, then the module does an insert 
> (another write operation).

I use Postgres, which as also returns the number of affected rows. Though
perhaps Kamailio doesn't support it for Postgres.


From spencer at 5ninesolutions.com  Thu Aug 30 00:39:08 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 29 Aug 2012 15:39:08 -0700
Subject: [SR-Users] Rtpproxy re-INVITE handling
Message-ID: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>

Hello,
I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a  
largely stock script.  Everything works great until the far end (on a public  
ip) sends a t.38 re-INVITE.  The 200OK from the NATed UAC then doesn't  
trigger rtpproxy and the private IP in the sdp causes the fax to fail.  Any  
help handling these re-INVITEs would be greatly appreciated.  I'm happy to  
post traces if that helps describe the situation.  The network topology  
looks like this:

NATed UAC -> Kamailio on a public IP -> UAS on a public IP  


Thanks in advance,
Spencer


Connected by DROID on Verizon Wireless

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From miconda at gmail.com  Thu Aug 30 09:51:04 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 09:51:04 +0200
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
Message-ID: <503F1B68.3060107@gmail.com>

Hello,

if your config it is based on the default one, the Route header for 
within dialog requests is marked by a parameter, nat=yes, that is used 
to decide whether to do rtpproxy or not.

So, if you have a custom config, check the default one for the nat 
traversal handling part.

Cheers,
Daniel

On 8/30/12 12:39 AM, Spencer Thomason wrote:
> Hello,
> I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in 
> a largely stock script.  Everything works great until the far end (on 
> a public ip) sends a t.38 re-INVITE.  The 200OK from the NATed UAC 
> then doesn't trigger rtpproxy and the private IP in the sdp causes the 
> fax to fail. Any help handling these re-INVITEs would be greatly 
> appreciated.  I'm happy to post traces if that helps describe the 
> situation.  The network topology looks like this:
>
> NATed UAC -> Kamailio on a public IP -> UAS on a public IP
>
>
> Thanks in advance,
> Spencer
>
>
> /Connected by DROID on Verizon Wireless/
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Thu Aug 30 09:56:20 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 09:56:20 +0200
Subject: [SR-Users] FW: Kamailio as sipRedirect
In-Reply-To: <3CB7D99A-D567-4FDF-A17B-B7718E554835@vanrise.com>
References: <04ee01cd85ea$8af4e880$a0deb980$@com> <503E5DB4.4040509@gmail.com>
	<3CB7D99A-D567-4FDF-A17B-B7718E554835@vanrise.com>
Message-ID: <503F1CA4.40103@gmail.com>

Hello,

if you know the IP address of 3CX, then you can skip authentication for 
that, like:

if(src_ip==__3cx_ip__) {
   # don't do authentication for the traffic coming here

}

You can use permission module to store the a list of ip addresses and 
then use functions from that module to match against source ip of 
requests, as alternative to above, providing more flexibility when 
dealing with lot of trusted ip addresses.

Cheers,
Daniel

On 8/29/12 9:15 PM, Fatima Chahrour wrote:
> Hello Danial,
>
> Hereby details:
> My current working 1st scenario is a call generated from A-party 
> x-lite registered on kamalio, this call is routed based on kamailio 
> lcr routing module which add prefix and send prefix to 3CX then 3CX 
> forward the call to B-party x-lite, registered on 3CX, successfully.
>
> In kamailio.cfg routing logic I only used :
>
> ...
> If (is_method("INVITE"))
> { if(!load_gateway("1")) {
> sl_send_reply("503", "unable to load gateways");
> Exit;  }
>
> { if(! next_gateway()) {
> sl_send_reply("503", "unable to find gateways");
> Exit;  }
>  Route(1);
> ....
>
> Now, my target is to register also the A-party x-lite to 3CX where I 
> added a rule in 3CX to forward the call comming to kamailio to take 
> the route desicion and act.
>
> So, here I need to know how can I tell kamailio to accept the call 
> from 3CX? this is what I meant by configuring 3CX on kamailio.
>
> I have weakness in the routing logic configuration, could you please 
> help in this too! What is the required replacement on the above to be 
> done?
>
> Moreover, I appreciate if you provide with docs or tutorials that can 
> help me understand more kamailio's routing logic functions!
>
> Much appreciated.
>
> Thanks in advance.
>
> F Chahrour
>
> Sent from iPhone
>
> On Aug 29, 2012, at 9:21 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>> Hello,
>>
>> On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote:
>>>
>>> Dears,
>>>
>>> I have an installed 3CX where I want to generate a call from it to 
>>> the Kamailo to get a list of suppliers to terminate the call to.
>>>
>>> Thus, the Kamailio should act as a sip redirect to the 3CX (for 
>>> routing decision).
>>>
>>> Have you tried such configuration before? And how do I configure the 
>>> 3CX on the Kamailio?
>>>
>> I don't understand what you expect from 'configure the 3CX on the 
>> Kamailio', can you add more details ?
>>
>> For sending a redirect, practically you have to update request uri 
>> and then send the 3xx reply, like:
>>
>> $ru = "sip:newnumber at newdomain.com";
>> send_reply("302", "Redirected";
>>
>> If you need to put more contacts in the redirect reply, then add:
>>
>> km_append_branch("uri");
>>
>> uri parameter can contain variables.
>>
>> Cheers,
>> Daniel
>>
>> -- 
>> Daniel-Constantin Mierla -http://www.asipto.com
>> http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Thu Aug 30 10:09:53 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 10:09:53 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120829191545.GJ75367@putsch.kolbu.ws>
References: <20120822163434.GH16374@putsch.kolbu.ws>
	<5035DBC6.9010909@gmail.com> <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
Message-ID: <503F1FD1.3090008@gmail.com>

Hello,

On 8/29/12 9:15 PM, ?yvind Kolbu wrote:
> On 2012-08-29 at 20:16, Daniel-Constantin Mierla wrote:
>> is usrloc module configured in db only mode?
> Yes, db_mode is 3.

this might not be very suitable for db_cluster usage if one db server is 
done, because right now the reading is done to the first available server.

>
>> Read operation cannot be done in 'parallel' fashion of the cluster, it
>> is only serial or round robin, stopping at first successful query (even
>> when there is no data records in the result of the query).
> Not related to this thread, but would it be possible to force it to try
> until data is found?

No, you have to do some C coding. The target was to offer high 
availability to access a database server, when one is down, try next.

Perhaps you can add some cross replication at database server layer.

>
>> If usrloc keeps also the records in memory, there is no select done at
>> runtime, only updates, which are write operations.
>>
>> db_check_update is only for SQL UPDATE command, in MySQL you can see the
>> number of affected rows and if it is 0, then the module does an insert
>> (another write operation).
> I use Postgres, which as also returns the number of affected rows. Though
> perhaps Kamailio doesn't support it for Postgres.

I am not a postgres user, but if it exports it, then it is used 
automatically by usrloc module.

>
>  From my previous post:
>
> DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7c9df08 (7) - [update [...]]
> DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0xb7cd2d90
> DEBUG: db_postgres [km_dbase.c:494]: 0xb7c9df08 PQresultStatus(PGRES_COMMAND_OK) PQgetResult(0x9058e78)
> DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x9058e78) result set
> DEBUG: <core> [db_res.c:81]: freeing 0 columns
> DEBUG: <core> [db_res.c:136]: freeing result set at 0xb7cd2d90
>
> but it did succed and updated one row, though not reflected in the logs.
>
>> So, the question is why do you think the check is done on one 'read
>> table', or you don't refer to the check of affected rows for UPDATE?
> Hm, perhaps I was a bit confused on how the affected rows works. Anyhow
> if a row is missing on a secondary server the UPDATE will fail and it
> will never be fixed with an INSERT.

IIRC, if the operation was successful, affected rows checks the last 
used connection. In a cluster, the operation is considered to be 
successful when int succeeded on a db server node.

I think you are looking more for replication, rather that high 
availability of the db layer. This is not done by db_cluster, if the 
nodes are up, then all should be fine, but if one goes down and then 
back up, its content is not synchronized, should be done by the db 
server itself.

Cheers,
Daniel

>
> Similar problem if the inital SELECT from the read-only sql-server returns
> noting, it will perform an INSERT on each of the write-servers, without
> first attempting to UPDATE. This will create duplicate contacts if the contact
> already exists on the write-servers. The stale entry will of course be
> pruned after a while..
>
> Below is from another experiment where I deleted the row in the location-table
> on the primary server, but left it on the secondary server:
>
> DEBUG: auth [api.c:218]: check_response: Authorization is OK
> [...]
> DEBUG: db_cluster [dbcl_api.c:436]: use table (voip_location) - cluster [data2]
> DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/0)
> DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/1)
> DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (8/0)
> DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (7/0)
> DEBUG: db_cluster [dbcl_api.c:299]: executing db cluster query command
> DEBUG: db_cluster [dbcl_api.c:300]: serial operation - cluster [data2] (8/0)
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 7 chars, out: 7 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
> DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7d03fd0 (7) - [select [... long statement ...]]
> [...]
> DEBUG: db_postgres [km_res.c:211]: no rows returned from the query
> DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x933fe78) result set
> DEBUG: usrloc [udomain.c:599]: aor 2549619 at hometest.voip.uio.no not found in table voip_location
> DEBUG: db_cluster [dbcl_api.c:284]: executing db cluster free-result command
> [...]
> DEBUG: <core> [sruid.c:176]: new sruid is [uloc-503e5fe0-26a8-4] (4 / 20)
> DEBUG: db_cluster [dbcl_api.c:436]: use table (voip_location) - cluster [data2]
> DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/0)
> DEBUG: db_cluster [dbcl_api.c:455]: set write table (voip_location) - cluster [data2] (9/1)
> DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (8/0)
> DEBUG: db_cluster [dbcl_api.c:445]: set read table (voip_location) - cluster [data2] (7/0)
> DEBUG: db_cluster [dbcl_api.c:333]: executing db cluster insert command
> DEBUG: db_cluster [dbcl_api.c:334]: parallel operation - cluster [data2] (9/0)
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 7 chars, out: 7 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 26 chars, out: 26 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 41 chars, out: 41 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 42 chars, out: 42 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 24 chars, out: 24 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
> DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7d03f08 (7) - [insert into voip_location [... long statment ..]]
> DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0xb7d076b8
> DEBUG: db_postgres [km_dbase.c:494]: 0xb7d03f08 PQresultStatus(PGRES_COMMAND_OK) PQgetResult(0x933fe78)
> DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x933fe78) result set
> DEBUG: <core> [db_res.c:81]: freeing 0 columns
> DEBUG: <core> [db_res.c:136]: freeing result set at 0xb7d076b8
> DEBUG: db_cluster [dbcl_api.c:334]: parallel operation - cluster [data2] (9/1)
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 7 chars, out: 7 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 26 chars, out: 26 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 41 chars, out: 41 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 42 chars, out: 42 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 24 chars, out: 24 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
> DEBUG: db_postgres [km_val.c:163]: PQescapeStringConn: in: 20 chars, out: 20 chars
> DEBUG: db_postgres [km_dbase.c:224]: sending query ok: 0xb7d03fd0 (7) - [insert into voip_location [ ... another blind insert ]]
> DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0xb7d076b8
> DEBUG: db_postgres [km_dbase.c:494]: 0xb7d03fd0 PQresultStatus(PGRES_COMMAND_OK) PQgetResult(0x933fe78)
> DEBUG: db_postgres [km_dbase.c:393]: PQclear(0x933fe78) result set
> DEBUG: <core> [db_res.c:81]: freeing 0 columns
> DEBUG: <core> [db_res.c:136]: freeing result set at 0xb7d076b8
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From oyvind.kolbu at usit.uio.no  Thu Aug 30 10:29:42 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Thu, 30 Aug 2012 10:29:42 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <503F1FD1.3090008@gmail.com>
References: <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
Message-ID: <20120830082942.GK75367@putsch.kolbu.ws>

On 2012-08-30 at 10:09, Daniel-Constantin Mierla wrote:
> On 8/29/12 9:15 PM, ?yvind Kolbu wrote:
> > On 2012-08-29 at 20:16, Daniel-Constantin Mierla wrote:
> >> is usrloc module configured in db only mode?
> > Yes, db_mode is 3.
> 
> this might not be very suitable for db_cluster usage if one db server is 
> done, because right now the reading is done to the first available server.

I choose 3 because of the usrloc documentation: ?The mode is useful if you
configure more servers sharing the same DB without any replication at SIP
level.? As I want multiple servers to write to the same database it seemed
as the best option.

> No, you have to do some C coding. The target was to offer high 
> availability to access a database server, when one is down, try next.

OK, will see if I get the time to do that.

> Perhaps you can add some cross replication at database server layer.

Yes, I've asked our DBAs for that, but they are not ready yet.

> > Hm, perhaps I was a bit confused on how the affected rows works. Anyhow
> > if a row is missing on a secondary server the UPDATE will fail and it
> > will never be fixed with an INSERT.
> 
> IIRC, if the operation was successful, affected rows checks the last 
> used connection. In a cluster, the operation is considered to be 
> successful when int succeeded on a db server node.
> 
> I think you are looking more for replication, rather that high 
> availability of the db layer. This is not done by db_cluster, if the 
> nodes are up, then all should be fine, but if one goes down and then 
> back up, its content is not synchronized, should be done by the db 
> server itself.

Yes, I actually want both, HA + replication, by using db_cluster as the
replicator. And as I've demonstrated the location-table on non-primary
servers will lack entries after downtime, which is bad.

If this indeed is impossible I've have to continue our current scheme with
SIP level replication.

-- 
?yvind Kolbu


From miconda at gmail.com  Thu Aug 30 11:32:18 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 11:32:18 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120830082942.GK75367@putsch.kolbu.ws>
References: <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
	<20120830082942.GK75367@putsch.kolbu.ws>
Message-ID: <503F3322.2060402@gmail.com>


On 8/30/12 10:29 AM, ?yvind Kolbu wrote:
> On 2012-08-30 at 10:09, Daniel-Constantin Mierla wrote:
>> On 8/29/12 9:15 PM, ?yvind Kolbu wrote:
>>> On 2012-08-29 at 20:16, Daniel-Constantin Mierla wrote:
>>>> is usrloc module configured in db only mode?
>>> Yes, db_mode is 3.
>> this might not be very suitable for db_cluster usage if one db server is
>> done, because right now the reading is done to the first available server.
> I choose 3 because of the usrloc documentation: ?The mode is useful if you
> configure more servers sharing the same DB without any replication at SIP
> level.? As I want multiple servers to write to the same database it seemed
> as the best option.

Yes, it is what it says, but you have two database servers. Your two 
kamailios don't connect to a single instance of DB server.

>
>> No, you have to do some C coding. The target was to offer high
>> availability to access a database server, when one is down, try next.
> OK, will see if I get the time to do that.
>
>> Perhaps you can add some cross replication at database server layer.
> Yes, I've asked our DBAs for that, but they are not ready yet.
>
>>> Hm, perhaps I was a bit confused on how the affected rows works. Anyhow
>>> if a row is missing on a secondary server the UPDATE will fail and it
>>> will never be fixed with an INSERT.
>> IIRC, if the operation was successful, affected rows checks the last
>> used connection. In a cluster, the operation is considered to be
>> successful when int succeeded on a db server node.
>>
>> I think you are looking more for replication, rather that high
>> availability of the db layer. This is not done by db_cluster, if the
>> nodes are up, then all should be fine, but if one goes down and then
>> back up, its content is not synchronized, should be done by the db
>> server itself.
> Yes, I actually want both, HA + replication, by using db_cluster as the
> replicator. And as I've demonstrated the location-table on non-primary
> servers will lack entries after downtime, which is bad.
Replication is done with parallel writing as long as all nodes are 
available, but, better said, you look for re-synchronization after 
downtime, which is another story, not easy to code, because it is db 
specific.

>
> If this indeed is impossible I've have to continue our current scheme with
> SIP level replication.
If you use db mode 3, then you can do database level replication, is the 
same.

Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From oyvind.kolbu at usit.uio.no  Thu Aug 30 12:50:54 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Thu, 30 Aug 2012 12:50:54 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <503F3322.2060402@gmail.com>
References: <20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
	<20120830082942.GK75367@putsch.kolbu.ws>
	<503F3322.2060402@gmail.com>
Message-ID: <20120830105054.GA8737@putsch.kolbu.ws>

On 2012-08-30 at 11:32, Daniel-Constantin Mierla wrote:
> On 8/30/12 10:29 AM, ?yvind Kolbu wrote:
> > Yes, I actually want both, HA + replication, by using db_cluster as the
> > replicator. And as I've demonstrated the location-table on non-primary
> > servers will lack entries after downtime, which is bad.
> Replication is done with parallel writing as long as all nodes are 
> available, but, better said, you look for re-synchronization after 
> downtime, which is another story, not easy to code, because it is db 
> specific.

I don't want Kamailio to synchronise my data, but I think it is reasonable
to expect it to treat the write targets individual and independent of the
result from the initial reading.

Perhaps enabling db_update_as_insert is the only current option.

> > If this indeed is impossible I've have to continue our current scheme with
> > SIP level replication.
> If you use db mode 3, then you can do database level replication, is the 
> same.

-------    -------
| DB1 |    | DB2 |
-------    -------
   |          |
   |          |
   |          |
-------   --------
| LOC1|   | LOC2 |
-------   --------
     \    /
   ----------
   | Phones |
   ----------

The above setup is my scenario. When everything is up LOC1 will use DB1 for
reading and write to both DB1 and DB2. Similarly LOC2 will use DB2 for
reading and write to both DB1 and DB2. Both uses the "other" DB as failover
for reading. LOC1 and LOC2 are setup with even load in an SRV record.

- While DB2 is down, say reboot for a new kernel, a phone A boots and
  registers at LOC1 and is populated in the DB1 database.  Reading works
  fine from LOC2 to DB1.
- DB2 is back again.
- Phone A re-registers at LOC1. The previous entry is found in the location
  table and an UPDATE is issued for both DB1 and DB2, but DB2 will still
  lack the entry.

DB2 will _never_ get an entry for until phone A boots and gets a new
Call-ID or for some reason phone A chooses to register with LOC2 instead.
Then a duplicate entry will end up in DB1, as LOC2 will blindly issue an
INSERT to both DB1 and DB2.

As the location servers are evenly used, ~every second call to phone A will
fail with 404.

-- 
?yvind Kolbu


From miconda at gmail.com  Thu Aug 30 13:32:48 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 13:32:48 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120830105054.GA8737@putsch.kolbu.ws>
References: <20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
	<20120830082942.GK75367@putsch.kolbu.ws>
	<503F3322.2060402@gmail.com>
	<20120830105054.GA8737@putsch.kolbu.ws>
Message-ID: <503F4F60.50308@gmail.com>


On 8/30/12 12:50 PM, ?yvind Kolbu wrote:
> On 2012-08-30 at 11:32, Daniel-Constantin Mierla wrote:
>> On 8/30/12 10:29 AM, ?yvind Kolbu wrote:
>>> Yes, I actually want both, HA + replication, by using db_cluster as the
>>> replicator. And as I've demonstrated the location-table on non-primary
>>> servers will lack entries after downtime, which is bad.
>> Replication is done with parallel writing as long as all nodes are
>> available, but, better said, you look for re-synchronization after
>> downtime, which is another story, not easy to code, because it is db
>> specific.
> I don't want Kamailio to synchronise my data, but I think it is reasonable
> to expect it to treat the write targets individual and independent of the
> result from the initial reading.
The write is independent of the read operation. Read cannot be performed 
in parallel, that results in duplicated records back to application.

Then, have in mind that there are several operations, one after the other:
- update - which do update to all write db servers
- test affected rows (which is always working on last write (update) 
connection)
- insert if affected rows is 0

So it is not an atomic operation, like 
updated_and_if_affected_rows_is_0_then_insert. All this layer is done in 
usrloc, in sequential steps, working fine for one server, but not for 
multiple nodes.

I am not sure what it will take to implement this kind of operation 
inside the database drivers, then it may work. TBased on quick thoughts, 
the code is there, just that has to be structured for each db connector 
and exported via db api and propagated to db_cluster.

>
> Perhaps enabling db_update_as_insert is the only current option.

This can end up in lot of records, as there is no update - if you set 
constraints at database layer, then you have failures on unique keys.

>
>>> If this indeed is impossible I've have to continue our current scheme with
>>> SIP level replication.
>> If you use db mode 3, then you can do database level replication, is the
>> same.
> -------    -------
> | DB1 |    | DB2 |
> -------    -------
>     |          |
>     |          |
>     |          |
> -------   --------
> | LOC1|   | LOC2 |
> -------   --------
>       \    /
>     ----------
>     | Phones |
>     ----------
>
> The above setup is my scenario. When everything is up LOC1 will use DB1 for
> reading and write to both DB1 and DB2. Similarly LOC2 will use DB2 for
> reading and write to both DB1 and DB2. Both uses the "other" DB as failover
> for reading. LOC1 and LOC2 are setup with even load in an SRV record.
>
> - While DB2 is down, say reboot for a new kernel, a phone A boots and
>    registers at LOC1 and is populated in the DB1 database.  Reading works
>    fine from LOC2 to DB1.
> - DB2 is back again.
> - Phone A re-registers at LOC1. The previous entry is found in the location
>    table and an UPDATE is issued for both DB1 and DB2, but DB2 will still
>    lack the entry.
>
> DB2 will _never_ get an entry for until phone A boots and gets a new
> Call-ID or for some reason phone A chooses to register with LOC2 instead.
> Then a duplicate entry will end up in DB1, as LOC2 will blindly issue an
> INSERT to both DB1 and DB2.
>
> As the location servers are evenly used, ~every second call to phone A will
> fail with 404.

You have to do cross replication at database layer and use db_cluster as 
read/write for failover access (e.g., try read/write on db1 and if 
fails, try the other one).

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From jh at tutpro.com  Thu Aug 30 14:43:02 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Thu, 30 Aug 2012 15:43:02 +0300
Subject: [SR-Users] db_cluster together with the registrar
	module	=	signal 11
In-Reply-To: <20120830082942.GK75367@putsch.kolbu.ws>
References: <50372BA5.5020003@gmail.com>
	<20120824084809.GI16374@putsch.kolbu.ws>
	<20120828163428.GC75367@putsch.kolbu.ws>
	<503D225C.4000304@gmail.com>
	<20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
	<20120830082942.GK75367@putsch.kolbu.ws>
Message-ID: <20543.24534.583438.350183@siika.tutpro.com>

?yvind Kolbu writes:

> Yes, I actually want both, HA + replication, by using db_cluster as the
> replicator. And as I've demonstrated the location-table on non-primary
> servers will lack entries after downtime, which is bad.

that should not be the case if you use master-master replication in your
db servers.

-- juha


From serusers at iptel.org  Thu Aug 30 15:21:51 2012
From: serusers at iptel.org (serusers at iptel.org)
Date: Thu, 30 Aug 2012 10:21:51 -0300
Subject: [SR-Users] Job ad - see details! Sent through  Search engine
Message-ID: <PP012U-9H3KKH-7C@uceajfgddcf.davisandsons.net>

Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  I would like to take this time to welcome you to our hiring
   process and give you a brief synopsis of the position's benefits and requirements.
   If you are taking a career break, are on a maternity leave, recently retired
   or simply looking for some part-time job, this position is for you. [...]
   

Content analysis details:   (6.7 points, 5.0 required)

 pts rule name              description
---- ---------------------- --------------------------------------------------
-0.0 SPF_PASS               SPF: sender matches SPF record
 1.4 RCVD_IN_BRBL_LASTEXT   RBL: RCVD_IN_BRBL_LASTEXT
                            [189.4.159.143 listed in bb.barracudacentral.org]
 0.4 RCVD_IN_XBL            RBL: Received via a relay in Spamhaus XBL
                            [189.4.159.143 listed in zen.spamhaus.org]
 3.3 RCVD_IN_PBL            RBL: Received via a relay in Spamhaus PBL
 1.3 RCVD_IN_RP_RNBL        RBL: Relay in RNBL,
                            https://senderscore.org/blacklistlookup/
                            [189.4.159.143 listed in bl.score.senderscore.com]
 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net
               [Blocked - see <http://www.spamcop.net/bl.shtml?189.4.159.143>]
-1.9 BAYES_00               BODY: Bayes spam probability is 0 to 1%
                            [score: 0.0000]
 0.8 RDNS_NONE              Delivered to internal network by a host with no rDNS


-------------- next part --------------
An embedded message was scrubbed...
From: <serusers at iptel.org>,
	<jiri at iptel.org>
Subject: Job ad - see details! Sent through  Search engine
Date: Thu, 30 Aug 2012 10:21:51 -0300
Size: 1994
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120830/aca4d276/attachment.eml>

From martian at centrum.sk  Thu Aug 30 15:52:05 2012
From: martian at centrum.sk (martian at centrum.sk)
Date: Thu, 30 Aug 2012 15:52:05 +0200
Subject: [SR-Users] Possible bug in authentication
In-Reply-To: <503C7502.2090609@pernau.at>
References: <20120823115419.63CC468C@centrum.sk>, <5036040F.3030908@gmail.com>,
	<20120823135149.427B12C2@centrum.sk>, <50362A69.8060807@pernau.at>,
	<20120824144154.3DAD39EA@centrum.sk> <503C7502.2090609@pernau.at>
Message-ID: <20120830155205.06BA2538@centrum.sk>

The ACK was indeed broken. The problem was at the SBC, where I did not expect it.
Everything works as it should.
Thank you very much for your help.
?
Martin
______________________________________________________________
> Od: "Klaus Darilion" 
> Komu: 
> D?tum: 28.08.2012 09:36
> Predmet: Re: [SR-Users] Possible bug in authentication
>
> CC: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List", miconda at gmail.com


On 24.08.2012 14:41, martian at centrum.sk wrote:
> The Route and Record-route headers are identical.
>
>
> ?From debug (when alias=domain.ch:5060):
>
> ----authentication of INVITE:
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: NOTICE: :
> ---------------------- In route(AUTH), just before
> from_uri==myself ----------------------
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==9 &&
> [domain.ch] == [127.0.0.1]
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==15 &&
> [domain.ch] == []
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==9 &&
> [domain.ch] == [127.0.0.1]
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:583]: grep_sock_info - checking if host==us: 10==15 &&
> [domain.ch] == []
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: DEBUG: 
> [socket_info.c:587]: grep_sock_info - checking if port 5060 (advertise
> 0) matches port 5060
>
> Aug 24 14:22:44 server /usr/sbin/kamailio[8588]: NOTICE: :
> ---------------------- from_uri==myself evaluated as
> TRUE!! ----------------------


Is this really a complete log? According to the log uri==myself should 
return FALSE as the compared strings are never the same.

> When I set alias=server.domain.ch:5060, from_uri==myself returns false
> (when determining if INVITE should be authenticated,resulting in
> replying 100 trying instead of 407 Proxy Auth Req) and loose_route()
> starts returning true and relays the ACK correctly.
>
> I can post more debug from this case also, but I didn't want to spam so
> much in one message.
>
> If you would like to see it, please let me know.
>
> So .. Shall I consider the loose_route() part fixed and assume that
> there MUST be a full name (hostname.domain:port) in the alias, when
> Kamailio is not used as a primary proxy for the domain?

No. It is rather simple: domain.ch is not identical to domain.ch:5060 
(as the first URI results in NAPTR+SRV lookups and my use another port 
than 5060).

Thus, if you want that Kamailio detects domain.ch as local domain, add 
"alias=domain.ch". If you want that Kamailio detects domain.ch:5060 as 
local domain add alias=domain.ch:5060 (not sure if quotes are needed here).

If you want that Kamailio accepts both domains as local domains, then 
add both alias.

Regardind loose_route: As Daniel mentioned, the ACK is broken.

regards
Klaus

>
> What about the from_uri==myself part?
>
> Martin
>
> ______________________________________________________________
> ?> Od: "Klaus Darilion" 
> ?> Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List" 
> ?> D?tum: 23.08.2012 15:04
> ?> Predmet: Re: [SR-Users] Possible bug in authentication
> ?>
>
> ?> CC: miconda at gmail.com
>
> The Route URI (sent by SBC) must be identical to the Record-Route URI
> (inserted by Kamailio).
>
> To find out why loose_route returns FALSE increase log-level.
> loose_route uses the "ismyself" function to evaluate if the Route header
> addresses this Kamailio server. And the "ismyself" is very verbose when
> doing this check.
>
> regards
> Klaus
>
> On 23.08.2012 13:51, martian at centrum.sk wrote:
> ?> Ok, so .. I have a session border controller device that is a contact
> ?> point for my SIP domain (SRV record in DNS set to its IP). All the
> ?> trafic goes through it and it does things like topology hiding etc.. The
> ?> device forwards the INVITE messages to Kamailio, because of the routing.
> ?>
> ?> The loose_route was working strangely, because it did not behave as
> ?> described in the documentation.
> ?>
> ?> Here is the sip message that it was suppose to pass:
> ?>
> ?> ACK sip:acc1 at domain.ch:5060 SIP/2.0
> ?>
> ?> Via: SIP/2.0/UDP domain.ch;branch=z9hG4bKac386033013
> ?>
> ?> Max-Forwards: 70
> ?>
> ?> From: "acc2" ;tag=1c1749458918
> ?>
> ?> To: ;tag=1c1892801634
> ?>
> ?> Call-ID: 17494024742382012111116@
> ?>
> ?> CSeq: 2 ACK
> ?>
> ?> Contact: 
> ?>
> ?> Route: 
> ?>
> ?> Supported: em,timer,replaces,path,resource-priority
> ?>
> ?> Allow:
> ?>
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
> ?>
> ?> User-Agent: SBC_DEVICE
> ?>
> ?> Content-Length: 0
> ?>
> ?> As you can see, there is a Route header and a To_tag .. so the
> ?> loose_route function should return true. But instead, it returned false,
> ?> then t_check_trans() also returned false and the routing logic exited
> ?> (exit;).
> ?>
> ?> This happens when the value of alias is not enclosed in double quotes.
> ?>
> ?> PS.: There is a "-" symbol in the domain name. Can't that be a problem
> ?> causing the need for the double quotes?
> ?>
> ?> PS2: Should there be only a domain name in the alias? or also the
> ?> hostname part? ... for example: ? domain.ch:5060 or server.domain.ch:5060
> ?>
> ?> Martin
> ?>
> ?> ______________________________________________________________
> ?> ?> Od: "Daniel-Constantin Mierla" 
> ?> ?> Komu: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> ?> Users Mailing List" 
> ?> ?> D?tum: 23.08.2012 12:21
> ?> ?> Predmet: Re: [SR-Users] Possible bug in authentication
> ?> ?>
> ?>
> ?> Hello,
> ?>
> ?> On 8/23/12 11:54 AM, martian at centrum.sk 
> wrote:
> ?>
> ?> ? ? Hello to everybody.
> ?>
> ?> ? ? I am currently working with Kamailio 3.3.1 on RedHat.
> ?>
> ?> ? ? The "loose_route" function was not working correctly and I observed
> ?> ? ? some very strange behaviour (not as one described in the
> ?> ? ? documentation of the function).
> ?>
> ?> ? ? I have found that there needs to be a port included in the "alias"
> ?> ? ? variable for the loose_route function to work correctly.
> ?>
> ?> ? ? However, upon adding the port to alias, the INVITE messages were no
> ?> ? ? longer authenticated (Kamailio just accepted them and didn't send
> ?> ? ? proxy-auth header in 407 message).
> ?>
> ?> ? ? My alias:
> ?>
> ?> ? ? alias="domain.ch:5060"
> ?>
> ?> ? ? Examining default routing logic, I found the problem here:
> ?>
> ?> ? ? if (is_method("REGISTER") || from_uri==myself)
> ?>
> ?> ? ? {
> ?>
> ?> ? ? # authenticate requests
> ?>
> ?> ? ? ...
> ?>
> ?> ? ? }
> ?>
> ?> ? ? The "from_uri==myself" was no longer evaluated as true, because
> ?> ? ? there was a port at the end of the alias.
> ?>
> ?> ? ? The FROM Header of the INVITE messages looks like:
> ?>
> ?> ? ? From: "acc1" ;tag=12345
> ?>
> ?> ? ? ..so .. no port number there.
> ?>
> ?> ? ? Btw, I have fixed this with replacing the "myself" list with my own
> ?> ? ? defined variable MY_DOMAIN.
> ?>
> ?> ? ? #!define MY_DOMAIN ".*@domain.ch" 
> ?>
> ?> ? ? So now the condition looks like this:
> ?>
> ?> ? ? if (is_method("REGISTER") || from_uri=~MY_DOMAIN)
> ?>
> ?> ? ? {
> ?>
> ?> ? ? ...
> ?>
> ?> ? ? }
> ?>
> ?> ? ? I am not sure if this is a bug that needs to be fixed or not. I am
> ?> ? ? just pointing my finger at it and I hope it will contribute to the
> ?> ? ? development.
> ?>
> ?> ? ? Also, a valid description of this behavior (when using port in
> ?> ? ? alias) would be appreciated.
> ?>
> ?>
> ?> if you enclose the value of the alias parameter in double quotes, then
> ?> it is taken as string value. If you want to set it to a host:port, then
> ?> remove the double quotes:
> ?>
> ?> alias=domain.ch:5060
> ?>
> ?>
> ?> Why do you say the loose_route() was working strangely? Do you add the
> ?> hostname as record-route, not the IP address? Detail more about what you
> ?> think is wrong with record routing/loose routing.
> ?>
> ?>
> ?> Cheers,
> ?> Daniel
> ?>
> ?> -- Daniel-Constantin Mierla
> -http://www.asipto.comhttp://twitter.com/#
> !/miconda
> ? -http://www.linkedin.com/in/micondaKamailio Advanced Training, Berlin,
> Nov 5-8, 2012 -http://asipto.com/u/kat
> ?>
> ?>
> ?>
> ?> _______________________________________________
> ?> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> ?> sr-users at lists.sip-router.org
> ?> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> ?>
>

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From oyvind.kolbu at usit.uio.no  Thu Aug 30 16:05:17 2012
From: oyvind.kolbu at usit.uio.no (=?utf-8?Q?=C3=98yvind?= Kolbu)
Date: Thu, 30 Aug 2012 16:05:17 +0200
Subject: [SR-Users] db_cluster together with the registrar module
	=	signal 11
In-Reply-To: <503F4F60.50308@gmail.com>
References: <20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
	<20120830082942.GK75367@putsch.kolbu.ws>
	<503F3322.2060402@gmail.com>
	<20120830105054.GA8737@putsch.kolbu.ws> <503F4F60.50308@gmail.com>
Message-ID: <20120830140517.GA24524@putsch.kolbu.ws>

On 2012-08-30 at 13:32, Daniel-Constantin Mierla wrote:
> Then, have in mind that there are several operations, one after the other:
> - update - which do update to all write db servers
> - test affected rows (which is always working on last write (update) 
> connection)
> - insert if affected rows is 0
> 
> So it is not an atomic operation, like 
> updated_and_if_affected_rows_is_0_then_insert. All this layer is done in 
> usrloc, in sequential steps, working fine for one server, but not for 
> multiple nodes.

After reading this mail and usrloc/ucontact.c I have a much better idea of
how this works, thanks!

> I am not sure what it will take to implement this kind of operation 
> inside the database drivers, then it may work. TBased on quick thoughts, 
> the code is there, just that has to be structured for each db connector 
> and exported via db api and propagated to db_cluster.

This would be great.

> You have to do cross replication at database layer and use db_cluster as 
> read/write for failover access (e.g., try read/write on db1 and if 
> fails, try the other one).

Yes, this is what we'll probably end up with. Thanks for all your answers!

-- 
?yvind Kolbu


From miconda at gmail.com  Thu Aug 30 16:16:37 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 16:16:37 +0200
Subject: [SR-Users] db_cluster together with the registrar module
 =	signal 11
In-Reply-To: <20120830140517.GA24524@putsch.kolbu.ws>
References: <20120828202333.GD75367@putsch.kolbu.ws>
	<503DAACC.5080808@gmail.com>
	<20120829112357.GE75367@putsch.kolbu.ws>
	<503E5C83.2050102@gmail.com>
	<20120829191545.GJ75367@putsch.kolbu.ws>
	<503F1FD1.3090008@gmail.com>
	<20120830082942.GK75367@putsch.kolbu.ws>
	<503F3322.2060402@gmail.com>
	<20120830105054.GA8737@putsch.kolbu.ws> <503F4F60.50308@gmail.com>
	<20120830140517.GA24524@putsch.kolbu.ws>
Message-ID: <503F75C5.6040700@gmail.com>


On 8/30/12 4:05 PM, ?yvind Kolbu wrote:
> On 2012-08-30 at 13:32, Daniel-Constantin Mierla wrote:
>> Then, have in mind that there are several operations, one after the other:
>> - update - which do update to all write db servers
>> - test affected rows (which is always working on last write (update)
>> connection)
>> - insert if affected rows is 0
>>
>> So it is not an atomic operation, like
>> updated_and_if_affected_rows_is_0_then_insert. All this layer is done in
>> usrloc, in sequential steps, working fine for one server, but not for
>> multiple nodes.
> After reading this mail and usrloc/ucontact.c I have a much better idea of
> how this works, thanks!
ok
>
>> I am not sure what it will take to implement this kind of operation
>> inside the database drivers, then it may work. TBased on quick thoughts,
>> the code is there, just that has to be structured for each db connector
>> and exported via db api and propagated to db_cluster.
> This would be great.
I agree, the good part is that all is open source, so anyone can jump in 
and contribute new features they need ;-)

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Thu Aug 30 16:38:17 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 16:38:17 +0200
Subject: [SR-Users] [OT] the role of SBCs
Message-ID: <503F7AD9.8040302@gmail.com>

Hello,

based on the outcome of the discussion carried in the thread:

http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html

I am looking again the clarify some aspects out there in the VoIP world.

So, as mainly dealing with proxy/sip signaling deployments, it's very 
often to be the first one hit by support issues claiming things don't 
work. Then you investigate and end up in conclusions like in the above 
thread:

"The problem was at the SBC, __where I did not expect it__."

The underlined part heats me up a bit, because I never understood from 
where it comes this perception that SBC is a MUST-TO-HAVE and the 
PERFECT (never mistaken or breaking things) node in a VoIP networks.

To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is 
there to break the things.

I don't want to start like a flame war, but is it something that I am 
obviously missing in regards to what benefits a SBC can bring? I see 
only inconveniences:
- another point of failure
- it is a b2bua, therefore very unlikely to offer the same performances 
of a proxy
- if transcoding is needed, a media server can be used behind the proxy, 
properly protected of attacks by the proxy and eventually deployed as a 
farm load balanced by the proxy
- if topology hinding is wanted in the b2bua fashion (not the proxy 
fashion with encoding headers), then the b2bua can be behind the proxy, 
properly protected of attacks by the proxy and eventually deployed as a 
farm load balanced by the proxy
- nat traversal was solved long time ago in proxy environment, being 
scalable by deploying a farm of rtp proxy

I don't want to go to other features, including the transport layer, 
it's a clear win of the proxy in my experience (ok, being deep involved 
in this project).

Then, what makes the SBC so desirable in many companies/voip 
deployments? If any SBC user here that can share, what was the reason to 
buy such a device? Any conceptual functionality that cannot be achieved 
with the proxy as the first hop in front of the (wild) clients?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From spencer at 5ninesolutions.com  Thu Aug 30 19:40:42 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 30 Aug 2012 10:40:42 -0700
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <503F1B68.3060107@gmail.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
Message-ID: <6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>

Hi Daniel,
Thanks for your help with this.  Here is a trace:
http://pastebin.com/pXeFbwBz

I see the nat=yes parameter added to the Route header.  

I've posted the script here:
http://pastebin.com/2qwHYHvj

Forgive my ignorance, I can't tell what I'm doing wrong.

Thanks!
Spencer


On Aug 30, 2012, at 12:51 AM, Daniel-Constantin Mierla wrote:

> Hello,
> 
> if your config it is based on the default one, the Route header for within dialog requests is marked by a parameter, nat=yes, that is used to decide whether to do rtpproxy or not.
> 
> So, if you have a custom config, check the default one for the nat traversal handling part.
> 
> Cheers,
> Daniel
> 
> On 8/30/12 12:39 AM, Spencer Thomason wrote:
>> Hello,
>> I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a largely stock script.  Everything works great until the far end (on a public ip) sends a t.38 re-INVITE.  The 200OK from the NATed UAC then doesn't trigger rtpproxy and the private IP in the sdp causes the fax to fail.  Any help handling these re-INVITEs would be greatly appreciated.  I'm happy to post traces if that helps describe the situation.  The network topology looks like this:
>> 
>> NATed UAC -> Kamailio on a public IP -> UAS on a public IP  
>> 
>> 
>> Thanks in advance,
>> Spencer
>> 
>> 
>> Connected by DROID on Verizon Wireless
>> 
>> 
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
> -- 
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From carsten at ng-voice.com  Thu Aug 30 20:26:28 2012
From: carsten at ng-voice.com (Carsten Bock)
Date: Thu, 30 Aug 2012 20:26:28 +0200
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <503F7AD9.8040302@gmail.com>
References: <503F7AD9.8040302@gmail.com>
Message-ID: <CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>

Hi Daniel,

i believe in many setups (including mine and the Sipwise Systems) a
SBC is always used as you described: Behind a Proxy.
The Proxy does Flood-detection, advanced logic, Loadbalancing and so
on and the SBC works only as simple B2B-UA for Transcoding and
Topology hiding.

Kind regards,
Carsten




2012/8/30 Daniel-Constantin Mierla <miconda at gmail.com>:
> Hello,
>
> based on the outcome of the discussion carried in the thread:
>
> http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html
>
> I am looking again the clarify some aspects out there in the VoIP world.
>
> So, as mainly dealing with proxy/sip signaling deployments, it's very often
> to be the first one hit by support issues claiming things don't work. Then
> you investigate and end up in conclusions like in the above thread:
>
> "The problem was at the SBC, __where I did not expect it__."
>
> The underlined part heats me up a bit, because I never understood from where
> it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never
> mistaken or breaking things) node in a VoIP networks.
>
> To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is
> there to break the things.
>
> I don't want to start like a flame war, but is it something that I am
> obviously missing in regards to what benefits a SBC can bring? I see only
> inconveniences:
> - another point of failure
> - it is a b2bua, therefore very unlikely to offer the same performances of a
> proxy
> - if transcoding is needed, a media server can be used behind the proxy,
> properly protected of attacks by the proxy and eventually deployed as a farm
> load balanced by the proxy
> - if topology hinding is wanted in the b2bua fashion (not the proxy fashion
> with encoding headers), then the b2bua can be behind the proxy, properly
> protected of attacks by the proxy and eventually deployed as a farm load
> balanced by the proxy
> - nat traversal was solved long time ago in proxy environment, being
> scalable by deploying a farm of rtp proxy
>
> I don't want to go to other features, including the transport layer, it's a
> clear win of the proxy in my experience (ok, being deep involved in this
> project).
>
> Then, what makes the SBC so desirable in many companies/voip deployments? If
> any SBC user here that can share, what was the reason to buy such a device?
> Any conceptual functionality that cannot be achieved with the proxy as the
> first hop in front of the (wild) clients?
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



-- 
Carsten Bock
CEO (Gesch?ftsf?hrer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:carsten at ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Gesch?ftsf?hrer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/


From miconda at gmail.com  Thu Aug 30 20:56:03 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 20:56:03 +0200
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
Message-ID: <503FB743.1000100@gmail.com>

Hello,

On 8/30/12 8:26 PM, Carsten Bock wrote:
> Hi Daniel,
>
> i believe in many setups (including mine and the Sipwise Systems) a
> SBC is always used as you described: Behind a Proxy.

yes, I do the same when I need a media server/b2bua for transcoding, etc

I'm trying to figure out what drives the other type of topology, with 
the SBC in front.

Of course, if anyone has good arguments against an front-end SBC, speak 
here as well, it may help other people to take the right decision. I 
pretty much listed mine in the initial message.

Cheers,
Daniel

> The Proxy does Flood-detection, advanced logic, Loadbalancing and so
> on and the SBC works only as simple B2B-UA for Transcoding and
> Topology hiding.
>
> Kind regards,
> Carsten
>
>
>
>
> 2012/8/30 Daniel-Constantin Mierla <miconda at gmail.com>:
>> Hello,
>>
>> based on the outcome of the discussion carried in the thread:
>>
>> http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html
>>
>> I am looking again the clarify some aspects out there in the VoIP world.
>>
>> So, as mainly dealing with proxy/sip signaling deployments, it's very often
>> to be the first one hit by support issues claiming things don't work. Then
>> you investigate and end up in conclusions like in the above thread:
>>
>> "The problem was at the SBC, __where I did not expect it__."
>>
>> The underlined part heats me up a bit, because I never understood from where
>> it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never
>> mistaken or breaking things) node in a VoIP networks.
>>
>> To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is
>> there to break the things.
>>
>> I don't want to start like a flame war, but is it something that I am
>> obviously missing in regards to what benefits a SBC can bring? I see only
>> inconveniences:
>> - another point of failure
>> - it is a b2bua, therefore very unlikely to offer the same performances of a
>> proxy
>> - if transcoding is needed, a media server can be used behind the proxy,
>> properly protected of attacks by the proxy and eventually deployed as a farm
>> load balanced by the proxy
>> - if topology hinding is wanted in the b2bua fashion (not the proxy fashion
>> with encoding headers), then the b2bua can be behind the proxy, properly
>> protected of attacks by the proxy and eventually deployed as a farm load
>> balanced by the proxy
>> - nat traversal was solved long time ago in proxy environment, being
>> scalable by deploying a farm of rtp proxy
>>
>> I don't want to go to other features, including the transport layer, it's a
>> clear win of the proxy in my experience (ok, being deep involved in this
>> project).
>>
>> Then, what makes the SBC so desirable in many companies/voip deployments? If
>> any SBC user here that can share, what was the reason to buy such a device?
>> Any conceptual functionality that cannot be achieved with the proxy as the
>> first hop in front of the (wild) clients?
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Thu Aug 30 20:59:35 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 30 Aug 2012 20:59:35 +0200
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
	<6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
Message-ID: <503FB817.4060009@gmail.com>

Hello,

I could not spot by quick eye checking what could happen there, the best 
is to use the debugger module with cfg_trace parameter set and check the 
execution trace to see what actions of the configuration file are 
executed and be sure the rtpproxy is called or not.

You can post the execution trace here if you need further help with it.

Cheers,
Daniel

On 8/30/12 7:40 PM, Spencer Thomason wrote:
> Hi Daniel,
> Thanks for your help with this.  Here is a trace:
> http://pastebin.com/pXeFbwBz
>
> I see the nat=yes parameter added to the Route header.
>
> I've posted the script here:
> http://pastebin.com/2qwHYHvj
>
> Forgive my ignorance, I can't tell what I'm doing wrong.
>
> Thanks!
> Spencer
>
>
> On Aug 30, 2012, at 12:51 AM, Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> if your config it is based on the default one, the Route header for 
>> within dialog requests is marked by a parameter, nat=yes, that is 
>> used to decide whether to do rtpproxy or not.
>>
>> So, if you have a custom config, check the default one for the nat 
>> traversal handling part.
>>
>> Cheers,
>> Daniel
>>
>> On 8/30/12 12:39 AM, Spencer Thomason wrote:
>>> Hello,
>>> I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() 
>>> in a largely stock script. Everything works great until the far end 
>>> (on a public ip) sends a t.38 re-INVITE.  The 200OK from the NATed 
>>> UAC then doesn't trigger rtpproxy and the private IP in the sdp 
>>> causes the fax to fail.  Any help handling these re-INVITEs would be 
>>> greatly appreciated.  I'm happy to post traces if that helps 
>>> describe the situation.  The network topology looks like this:
>>>
>>> NATed UAC -> Kamailio on a public IP -> UAS on a public IP
>>>
>>>
>>> Thanks in advance,
>>> Spencer
>>>
>>>
>>> /Connected by DROID on Verizon Wireless/
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>> -- 
>> Daniel-Constantin Mierla -http://www.asipto.com
>> http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>

-- 
Daniel-Constantin Mierla -http://www.asipto.com
http://twitter.com/#!/miconda  -http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat

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From rupert.organ at bt.com  Thu Aug 30 21:58:08 2012
From: rupert.organ at bt.com (rupert.organ at bt.com)
Date: Thu, 30 Aug 2012 20:58:08 +0100
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <503FB743.1000100@gmail.com>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>,
	<503FB743.1000100@gmail.com>
Message-ID: <8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>

Hello,
I have only ever deployed SER behind SBC. SBC is the most expensive component. I am not involved with security, but I would not be allowed by my organisation to deploy something like SER, with mysql details of usernames and password on an internet address. The SBC hides my addressing and only opens temp pinholes for both sip and rtp. The main USP of the SBCs is the power to route many rtp streams simultaneously. This also allows guard timers to check rtp streams dying so no overcharging takes place. I don't think the hardware on a conventional server will scale to the number of rtp streams i want to deal with. I guess u might say, why route all the rtps through a single  point of failure... there are other mechanisms to avoid overcharging,and also  can hide SER behind NAT... I agree, but SBC suppliers have moved along the value chain....header manipulation for interworking, eNum lookup, codec manipulations....again all things that can be done on SER.... it comes down to a philosophy and "way we do things"...The other main thing SBC does is prevent attacks. i.e. SIP signalling attacks result in instant filtering from that address and thereby protects the service. I wouldn't know how to protect my SER if it was exposed to the internet, ...maybe there is a way...I just don't know it.
Rupert.


________________________________________
From: sr-users-bounces at lists.sip-router.org [sr-users-bounces at lists.sip-router.org] On Behalf Of Daniel-Constantin Mierla [miconda at gmail.com]
Sent: 30 August 2012 19:56
To: Carsten Bock
Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users        Mailing List
Subject: Re: [SR-Users] [OT] the role of SBCs

Hello,

On 8/30/12 8:26 PM, Carsten Bock wrote:
> Hi Daniel,
>
> i believe in many setups (including mine and the Sipwise Systems) a
> SBC is always used as you described: Behind a Proxy.

yes, I do the same when I need a media server/b2bua for transcoding, etc

I'm trying to figure out what drives the other type of topology, with
the SBC in front.

Of course, if anyone has good arguments against an front-end SBC, speak
here as well, it may help other people to take the right decision. I
pretty much listed mine in the initial message.

Cheers,
Daniel

> The Proxy does Flood-detection, advanced logic, Loadbalancing and so
> on and the SBC works only as simple B2B-UA for Transcoding and
> Topology hiding.
>
> Kind regards,
> Carsten
>
>
>
>
> 2012/8/30 Daniel-Constantin Mierla <miconda at gmail.com>:
>> Hello,
>>
>> based on the outcome of the discussion carried in the thread:
>>
>> http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html
>>
>> I am looking again the clarify some aspects out there in the VoIP world.
>>
>> So, as mainly dealing with proxy/sip signaling deployments, it's very often
>> to be the first one hit by support issues claiming things don't work. Then
>> you investigate and end up in conclusions like in the above thread:
>>
>> "The problem was at the SBC, __where I did not expect it__."
>>
>> The underlined part heats me up a bit, because I never understood from where
>> it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never
>> mistaken or breaking things) node in a VoIP networks.
>>
>> To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is
>> there to break the things.
>>
>> I don't want to start like a flame war, but is it something that I am
>> obviously missing in regards to what benefits a SBC can bring? I see only
>> inconveniences:
>> - another point of failure
>> - it is a b2bua, therefore very unlikely to offer the same performances of a
>> proxy
>> - if transcoding is needed, a media server can be used behind the proxy,
>> properly protected of attacks by the proxy and eventually deployed as a farm
>> load balanced by the proxy
>> - if topology hinding is wanted in the b2bua fashion (not the proxy fashion
>> with encoding headers), then the b2bua can be behind the proxy, properly
>> protected of attacks by the proxy and eventually deployed as a farm load
>> balanced by the proxy
>> - nat traversal was solved long time ago in proxy environment, being
>> scalable by deploying a farm of rtp proxy
>>
>> I don't want to go to other features, including the transport layer, it's a
>> clear win of the proxy in my experience (ok, being deep involved in this
>> project).
>>
>> Then, what makes the SBC so desirable in many companies/voip deployments? If
>> any SBC user here that can share, what was the reason to buy such a device?
>> Any conceptual functionality that cannot be achieved with the proxy as the
>> first hop in front of the (wild) clients?
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>

--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat


_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

From agranig at sipwise.com  Fri Aug 31 00:28:42 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 31 Aug 2012 00:28:42 +0200
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>,
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
Message-ID: <503FE91A.6010503@sipwise.com>

Hi Rupert,

The interesting thing is that the arguments you brought up are exactly
the same in every big corporation we've worked with. They're perfectly
valid, also taking into account the decision making process, which
department takes responsibility for what, and it's also of course
politically and policy driven.

Still, all of it can be solved quite easily with proper sip proxies and
b2buas and efficient mediaproxies, see below.

On 08/30/2012 09:58 PM, rupert.organ at bt.com wrote:
> I have only ever deployed SER behind SBC. SBC is the most expensive component. 

And there, the thinking often goes like "it's the most expensive
component, so it must be the most secure one", and nobody got fired for
buying an Acme SBC so far. SBCs are marketed as "firewalls for VoIP",
and security people love this idea.

> I am not involved with security, but I would not be allowed by my organisation to deploy something like SER, with mysql details of usernames and password on an internet address. 

In our systems we've a stateless kamailio instance running as load
balancer as single ingress/egress point in front of the proxies, b2buas,
app servers etc, and it only has a very small dbtext db for the
dispatcher table. No credentials of any kind exposed, and you can put it
on a separate node if you want.

> The SBC hides my addressing and only opens temp pinholes for both sip and rtp. The main USP of the SBCs is the power to route many rtp streams simultaneously. This also allows guard timers to check rtp streams dying so no overcharging takes place. I don't think the hardware on a conventional server will scale to the number of rtp streams i want to deal with. 

Our mediaproxy tells the b2bua internally via XMLRPC to tear down a call
if all streams of a call time out. Media relaying is done directly in
the kernel, you can handle thousands of parallel calls on a conventional
machine with a proper NIC, and you can add as many new nodes as you
like/need.

>I guess u might say, why route all the rtps through a single  point of failure... there are other mechanisms to avoid overcharging,and also  can hide SER behind NAT... I agree, but SBC suppliers have moved along the value chain....header manipulation for interworking, eNum lookup, codec manipulations....again all things that can be done on SER.... it comes down to a philosophy and "way we do things"...The other main thing SBC does is prevent attacks. i.e. SIP signalling attacks result in instant filtering from that address and thereby protects the service. I wouldn't know how to protect my SER if it was exposed to the internet, ...maybe there is a way...I just don't know it.

We do NAT handling on the kamailio lb instance, we do header
manipulation and codec filtering on an open source sbc (sems), and we do
DoS/DDoS protection also in kamailio lb, quite similar as described
here: http://kb.asipto.com/kamailio:usage:k31-sip-scanning-attack

The reason for us to use SBCs is really just to provide B2BUA features
like doing session timer handling, header/codec filtering, clean
topology hiding, and media stuff like music-on-hold and transcoding
etc., which all has been mentioned in different threads.

We still have customers who put an Acme SBC in front of all this, and
well, if they're comfortable with this and feel better, it's perfectly
fine. It more sometimes turns out though that the SBC behaves
incorrectly in quite a few scenarios, and then time-to-fix is most much
faster in an open source system rather than an expensive SBC appliance.

Andreas

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From spencer at 5ninesolutions.com  Fri Aug 31 03:41:34 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 30 Aug 2012 18:41:34 -0700
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <503FB817.4060009@gmail.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
	<6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
	<503FB817.4060009@gmail.com>
Message-ID: <B03A90C3-6658-4F80-94BE-7EBB7D39287E@5ninesolutions.com>

Hi Daniel,
I can confirm that rtpproxy_manage is called. See:
http://pastebin.com/ZVXjK9ry

I'm seeing ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy in the logs when processing the 200OK in the re-INVITE.  I've included a debug level log from rtpproxy in the log as well. 

When handling the re-INVITE there is:
	? Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command "25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 184.170.249.3 32122 3ae1Dvgr5vmeg;1 199857477;1"
	? Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: adding strong flag to existing session, new=1/0/0
	? Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup on ports 55324/46010, session timer restarted
	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply "25778_11 46010 184.170.249.8#012"

but the 200OK:
	? Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command "25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 71.104.248.48 6016 3ae1Dvgr5vmeg;1"
	? Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup request failed: session 1952045641-6076-15 at BA.FJ.B.CFC, tags 3ae1Dvgr5vmeg;1/NONE not found
	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply "25778_12 0 184.170.249.8#012"
	? Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy

I'm not familiar with the rtpproxy commands to know why it cannot locate the session.

Thanks for your assistance,
Spencer


On Aug 30, 2012, at 11:59 AM, Daniel-Constantin Mierla wrote:

> Hello,
> 
> I could not spot by quick eye checking what could happen there, the best is to use the debugger module with cfg_trace parameter set and check the execution trace to see what actions of the configuration file are executed and be sure the rtpproxy is called or not.
> 
> You can post the execution trace here if you need further help with it.
> 
> Cheers,
> Daniel
> 
> On 8/30/12 7:40 PM, Spencer Thomason wrote:
>> Hi Daniel,
>> Thanks for your help with this.  Here is a trace:
>> http://pastebin.com/pXeFbwBz
>> 
>> I see the nat=yes parameter added to the Route header.  
>> 
>> I've posted the script here:
>> http://pastebin.com/2qwHYHvj
>> 
>> Forgive my ignorance, I can't tell what I'm doing wrong.
>> 
>> Thanks!
>> Spencer
>> 
>> 
>> On Aug 30, 2012, at 12:51 AM, Daniel-Constantin Mierla wrote:
>> 
>>> Hello,
>>> 
>>> if your config it is based on the default one, the Route header for within dialog requests is marked by a parameter, nat=yes, that is used to decide whether to do rtpproxy or not.
>>> 
>>> So, if you have a custom config, check the default one for the nat traversal handling part.
>>> 
>>> Cheers,
>>> Daniel
>>> 
>>> On 8/30/12 12:39 AM, Spencer Thomason wrote:
>>>> Hello,
>>>> I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a largely stock script.  Everything works great until the far end (on a public ip) sends a t.38 re-INVITE.  The 200OK from the NATed UAC then doesn't trigger rtpproxy and the private IP in the sdp causes the fax to fail.  Any help handling these re-INVITEs would be greatly appreciated.  I'm happy to post traces if that helps describe the situation.  The network topology looks like this:
>>>> 
>>>> NATed UAC -> Kamailio on a public IP -> UAS on a public IP  
>>>> 
>>>> 
>>>> Thanks in advance,
>>>> Spencer
>>>> 
>>>> 
>>>> Connected by DROID on Verizon Wireless
>>>> 
>>>> 
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> 
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>> 
>>> -- 
>>> Daniel-Constantin Mierla - 
>>> http://www.asipto.com
>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> 
>>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - 
>>> http://asipto.com/u/kat
>> 
> 
> -- 
> Daniel-Constantin Mierla - 
> http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> 
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - 
> http://asipto.com/u/kat



From patrice.bodeven at orange.com  Thu Aug 30 10:43:14 2012
From: patrice.bodeven at orange.com (patrice.bodeven at orange.com)
Date: Thu, 30 Aug 2012 08:43:14 +0000
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <5035C662.9000203@pernau.at>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
Message-ID: <3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>

Hello, 

Sorry, but i am blocked on the nonce reuse between the Register and the INVITE even with the info provided in debug mode. 
 
My understanding is Nonce should be valid between SIP methods until the nonce is valid in the time. 
But I don't know how the Nonce is stored and how Kamailio is looking for the nonce into the memory ! 

I did a test on Register => nonce reuse is ok. 
I did a test on INVITE => nonce reuse of Register or on previous INVITE doesn't work !   

But based on the site indicated below and the SIPP (generate the INVITE with the Nonce of Register), the result is 
auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
auth [api.c:147]: auth:pre_auth: Invalid nonce value received

But there is no explanation why the nonce is invalid.
 
See the wireshark trace : the nonce into the Invite is the same as the nonce used in the Register. 

Thank you for your help. 

Note -

One_time_nonce is set at default value.
 
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 2 contacts per AOR */
modparam("registrar", "max_contacts", 2)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# remove the sip from the realm
modparam("registrar", "realm_prefix", "sip:")
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURLAUTH)
modparam("usrloc", "db_mode", 0)
modparam("usrloc", "use_domain", MULTIDOMAIN)
modparam("usrloc", "hash_size", 15) # size of table 30k users
modparam("usrloc", "timer_interval", 180) # Every 180s request to DB to delete expired contacts
#!endif

#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURLAUTH)
modparam("auth", "auth_checks_register", 6) # callid and From TAG
modparam("auth", "qop", "auth")    # enable qop=auth
modparam("auth", "nonce_count", 1) # enable nonce_count support 
#modparam("auth", "nc_array_order", 20)   # 2 puissance 20 - 1M in-flight nonces, using 1Mb memory
#modparam("auth", "nid_pool_no", 4)
modparam("auth", "nonce_expire", 21600)   # Set nonce_expire to 6 hours 
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN) (multidomain is 0)

        # authenticate the REGISTER requests (uncomment to enable auth)
                if (!www_authorize("$td", "subscriber"))
                {
          if($? == -2 || $? == -3) {
            sl_send_reply("403","Forbidden auth ID");
            exit;
          }
          else {
                        www_challenge("$td", "17");
                            exit;
                       }
            }

Sur invite : 
          if (!proxy_authorize("$fd", "subscriber")) {
           if($? == -2 || $? == -3) {
              sl_send_reply("403","Forbidden auth ID");
              exit;
            }
            else {
                                proxy_challenge("$fd", "17");
                xlog("L_INFO","INVITE challenged by 407 to :$di:$dp \n");
                                exit;
                        }
            }
                        if (is_method("PUBLISH"))
                        {
                                if ($au!=$fU || $au!=$tU) {
                                        sl_send_reply("403","Forbidden auth ID");
                                        exit;
                                }
                                if ($au!=$rU) {
                                        sl_send_reply("403","Forbidden R-URI");
                                        exit;
                                }
#!ifdef WITH_MULTIDOMAIN
                                if ($fd!=$rd) {
                                        sl_send_reply("403","Forbidden R-URI domain");
                                        exit;
                                }
#!endif
                        }
Cordialement/ Best regards 
Patrice B

-----Message d'origine-----
De?: Klaus Darilion [mailto:klaus.mailinglists at pernau.at] 
Envoy??: jeudi 23 ao?t 2012 07:58
??: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
Cc?: BODEVEN Patrice RD-CORE
Objet?: Re: [SR-Users] Kamailio - Nonce validity between Register and Invite

The nonce is globally valid until it expires (for all kind of requests, 
IIRC you could also change the user).

Maybe your "internal" SIP clients calculates the response wrong. You 
could test the response calculation with this website (do not use 'real' 
passwords):
http://pernau.at/kd/sipdigest.php

klaus

On 22.08.2012 10:51, patrice.bodeven at orange.com wrote:
> Hello,
>
> I am working on Kamailio 3.2.2. There is no traffic, only functional
> test done.
>
> Based on the SIP Client used until now (Xlite), the INVITE is
> systematically authenticated by 407 as there is no Proxy-Authorization
> in the initial INVITE. Expected/normal behavior.
>
> But when using an internal SIP client, the initial INVITE is generated
> with a Proxy-Authorization containing the nonce value used by the
> successful registration.
>
> _In Authorization of Register (the answer is 200OK)_
>
> Digest username="+33296488922 at ims.v0.pftest.net",
> realm="sip:ims.v0.pftest.net",
> nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA",
> uri="sip:ims.v0.pftest.net",
> response="4ed51b8e62cf769b19f8c4771e068ad3",
> cnonce="06440000000529330fe1", qop=auth, nc=00000011
>
> _In Proxy-Authorization of Invite _
>
> Digest username="+33296488922 at ims.v0.pftest.net",
> realm="sip:ims.v0.pftest.net",
> nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA",
> uri="sip:0157361149 at ims.v0.pftest.net",
> response="b247052d6d5b37e0b6677c816390cb77",
> cnonce="06440000000529330fe1", qop=auth, nc=00000012
>
> As the nonce expire is high (see nonce_expire), I don't understand why
> Kamailio requests the Authentication of the INVITE by 407.
>
> _407 with Proxy-Authenticate_: Digest realm="ims.v0.pftest.net",
> nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true
>
> *Does it mean that the REGISTER and INVITE are using different Nonce
> value ? *
>
> ?Information on my configuration file.
>
> modparam("auth_db", "db_url", DBURLAUTH)
>
> modparam("auth", "auth_checks_register", 6) # callid and From TAG
>
> modparam("auth", "qop", "auth")    # enable qop=auth
>
> modparam("auth", "nonce_count", 1) # enable nonce_count support
>
> modparam("auth", "nonce_expire", 21600)   # Set nonce_expire to 6 hours
>
> For register, I do
>
>                   if (!www_authorize("$ru", "subscriber"))
>
>                  {
>
>                     if($? == -2 || $? == -3) {
>
>                        sl_send_reply("403","Forbidden auth ID");
>
>                       exit;
>
>                     }
>
>                     else {
>
>                          www_challenge("$ru", "17");
>
>                          exit;
>
>                     }
>
>                  }
>
> For invite, I do
>
>             if (!proxy_authorize("$fd", "subscriber"))
>
>             {
>
>                 if($? == -2 || $? == -3) {
>
>                    sl_send_reply("403","Forbidden auth ID");
>
>                    exit;
>
>                 }
>
>                else {
>
>                     proxy_challenge("$fd", "17");
>
>                     xlog("L_INFO","INVITE challenged by 407 to :$di:$dp
> \n");
>
>                     exit;
>
>                  }
>
>             }
>
> Followed by consume_credentials();
>
> Thanks in advance for your support.
>
> *Patrice B*
>
> _________________________________________________________________________________________________________________________
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>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>

_________________________________________________________________________________________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.

This message and its attachments may contain confidential or privileged information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete this message and its attachments.
As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified.
Thank you.

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From nordseven at gmail.com  Fri Aug 31 09:44:01 2012
From: nordseven at gmail.com (Nord7)
Date: Fri, 31 Aug 2012 11:44:01 +0400
Subject: [SR-Users] LCR (next_gw) question
Message-ID: <CAHViUgd_VUB_5YAv7_WyYMMnxkDBhrT5KXt1dZ8q-q7hQ-=5vQ@mail.gmail.com>

Hello everyone. Configures the module LCR. I can't get him to take the next
on the list of priorities gateway if the first does not work. Function
"next_gw" capable of doing that? Did anyone realties that? If possible show
example.

Kamailio 3.3

Regards, Kirill
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From oej at edvina.net  Fri Aug 31 09:47:02 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Fri, 31 Aug 2012 09:47:02 +0200
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <503FE91A.6010503@sipwise.com>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>,
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
Message-ID: <FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>

Good discussion!

In most, but not all, cases it's a political/business decision outside of the scope of the technichal specifications. A commercial SBC delivers a cloud of magic dust that makes some people feel better and more secure. I have audited several SBC installations that are totally insecure, where the local techies lack knowledge on how to operate it. Management people think the SBC is secure by design. I can't blame the vendors here - it's more correct to blame the decision process.

You can always build an SBC with Kamailio plus a media server like sems, Asterisk or FreeSwitch in combination with IPtables. It requires experience and knowledge, but it is propably a better longterm to invest in local knowledge for security instead of relying on outside companies without building local knowledge...

Another problem is the architecture issue. SIP took off as a PSTN replacement. The design of the SIP network and a traditional switch/PBX network is radically different, so SIP networks was built with non-SIP design, with heavy media servers (b2bua) controlling every call. This put a severe limitation in the service set and forced a big dependency into the design, that wasn't there in the original SIP design. In order for SIP to get anywhere beyond G711 calls we need to rip that apart.

Trying to do that is hard, since the endpoints are more and more designed for this design. Normal phone services like music-on-hold and call parking are hard to deliver without support from the phones. That forces many design to move back to a PBX design. 

It takes time to change people's mindset. We need to prove that we can build better and interoperable solutions. I can't get SIP presence to work across vendors. That's bad for the industry, bad for us and hurts the trust in new designs and solutions. Which moves people back to the old trustworthy and reliant PBX vendor. And the SBC, because that's what gartner group and similar institutes tell them is the proper design.

The battle isn't over. We need to prove our vision. Andreas and SIPwise is doing great work here, as is AG Projects. And, of course, the whole Kamailio community and excellent developer team.

/O

* The new Edvina SIP Masterclass - Stockholm Oct, Miami, FL Dec 2012
  http://edvina.net/training/new-sip-masterclass/
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From miconda at gmail.com  Fri Aug 31 10:34:40 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 10:34:40 +0200
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
	<3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
Message-ID: <50407720.8070109@gmail.com>

Hello,

On 8/30/12 10:43 AM, patrice.bodeven at orange.com wrote:
> Hello,
>
> Sorry, but i am blocked on the nonce reuse between the Register and the INVITE even with the info provided in debug mode.
>   
> My understanding is Nonce should be valid between SIP methods until the nonce is valid in the time.
> But I don't know how the Nonce is stored and how Kamailio is looking for the nonce into the memory !
>
> I did a test on Register => nonce reuse is ok.
> I did a test on INVITE => nonce reuse of Register or on previous INVITE doesn't work !
>
> But based on the site indicated below and the SIPP (generate the INVITE with the Nonce of Register), the result is
> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>
> But there is no explanation why the nonce is invalid.
can you apply the following patch:

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/auth/api.c;h=c79de5c9dd6cd3e176c05d836c7b0a4275f8a3d2;hp=2ee39a7a6cbcce741365dac97ae8a3db8c196b0c;hb=494b383edde7a2d193c220f3117506e4cc95932f;hpb=e4ecf49add0e62330e1db071106806e35e9b078a

It practically adds the return code in the log message, that should give 
better leads to why the nonce is considered invalid, allowing to track 
back quicker in the source code.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Fri Aug 31 10:46:23 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 10:46:23 +0200
Subject: [SR-Users] spam messages breaking through filters
In-Reply-To: <503B1A0F.8050503@gmail.com>
References: <503B1A0F.8050503@gmail.com>
Message-ID: <504079DF.1030204@gmail.com>

Hello,

after updating the filters on mailing list manager to hold everything 
marked at list with:

X-Spam-Level: *

there were two other spam messages, coming with:

X-Spam-Level:

like:

http://lists.sip-router.org/pipermail/sr-users/2012-August/074479.html

The content has the spam message, but that happens because there are two 
spamassassin checks, one done at iptel.org and one done by 
lists.sip-router.org (hosted by kamailio.org). For some reason, 
spamassassin on kamailio.org does not detects it as spam anymore, 
setting X-Spam-Level to empty.

Is anyone familiar with such setup of chaining spamassassin? Is there a 
setting we miss in spamassasin to tell to check also X-Spam-Level of 
incoming message and trust it, eventually when coming from a list of 
preset IP addresses.

Thanks,
Daniel

On 8/27/12 8:56 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> there were some spam messages going through our mailing list filters. 
> I was updating the filters on mailman, but there are new and new ideas 
> coming from this guys.
>
> Sorry for the pollution and I wanted to say that these messages are 
> marked as spam by spamassasin, you can filter them locally as well 
> looking for the header
>
> X-Spam-Level: ***
>
> You can adjust the number of '*'.
>
> Some email services do that automatically, like gmail.com for example.
>
> Of course, if anyone has better ideas of protection, please write here.
>
> Cheers,
> Daniel
>


-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Fri Aug 31 10:53:32 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 10:53:32 +0200
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <B03A90C3-6658-4F80-94BE-7EBB7D39287E@5ninesolutions.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
	<6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
	<503FB817.4060009@gmail.com>
	<B03A90C3-6658-4F80-94BE-7EBB7D39287E@5ninesolutions.com>
Message-ID: <50407B8C.6090503@gmail.com>

Hello,

On 8/31/12 3:41 AM, Spencer Thomason wrote:
> Hi Daniel,
> I can confirm that rtpproxy_manage is called. See:
> http://pastebin.com/ZVXjK9ry
>
> I'm seeing ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy in the logs when processing the 200OK in the re-INVITE.  I've included a debug level log from rtpproxy in the log as well.
this can happen because the rtpproxy was not engaged for the request, 
but only for the reply.

As you say, the logs are for the 200OK, what about the ones for request, 
is rtpproxy called there?

Cheers,
Daniel

>
> When handling the re-INVITE there is:
> 	? Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command "25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 184.170.249.3 32122 3ae1Dvgr5vmeg;1 199857477;1"
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: adding strong flag to existing session, new=1/0/0
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup on ports 55324/46010, session timer restarted
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply "25778_11 46010 184.170.249.8#012"
>
> but the 200OK:
> 	? Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command "25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 71.104.248.48 6016 3ae1Dvgr5vmeg;1"
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup request failed: session 1952045641-6076-15 at BA.FJ.B.CFC, tags 3ae1Dvgr5vmeg;1/NONE not found
> 	? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply "25778_12 0 184.170.249.8#012"
> 	? Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy
>
> I'm not familiar with the rtpproxy commands to know why it cannot locate the session.
>
> Thanks for your assistance,
> Spencer
>
>
> On Aug 30, 2012, at 11:59 AM, Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> I could not spot by quick eye checking what could happen there, the best is to use the debugger module with cfg_trace parameter set and check the execution trace to see what actions of the configuration file are executed and be sure the rtpproxy is called or not.
>>
>> You can post the execution trace here if you need further help with it.
>>
>> Cheers,
>> Daniel
>>
>> On 8/30/12 7:40 PM, Spencer Thomason wrote:
>>> Hi Daniel,
>>> Thanks for your help with this.  Here is a trace:
>>> http://pastebin.com/pXeFbwBz
>>>
>>> I see the nat=yes parameter added to the Route header.
>>>
>>> I've posted the script here:
>>> http://pastebin.com/2qwHYHvj
>>>
>>> Forgive my ignorance, I can't tell what I'm doing wrong.
>>>
>>> Thanks!
>>> Spencer
>>>
>>>
>>> On Aug 30, 2012, at 12:51 AM, Daniel-Constantin Mierla wrote:
>>>
>>>> Hello,
>>>>
>>>> if your config it is based on the default one, the Route header for within dialog requests is marked by a parameter, nat=yes, that is used to decide whether to do rtpproxy or not.
>>>>
>>>> So, if you have a custom config, check the default one for the nat traversal handling part.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/30/12 12:39 AM, Spencer Thomason wrote:
>>>>> Hello,
>>>>> I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a largely stock script.  Everything works great until the far end (on a public ip) sends a t.38 re-INVITE.  The 200OK from the NATed UAC then doesn't trigger rtpproxy and the private IP in the sdp causes the fax to fail.  Any help handling these re-INVITEs would be greatly appreciated.  I'm happy to post traces if that helps describe the situation.  The network topology looks like this:
>>>>>
>>>>> NATed UAC -> Kamailio on a public IP -> UAS on a public IP
>>>>>
>>>>>
>>>>> Thanks in advance,
>>>>> Spencer
>>>>>
>>>>>
>>>>> Connected by DROID on Verizon Wireless
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>>
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>> -- 
>>>> Daniel-Constantin Mierla -
>>>> http://www.asipto.com
>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>>
>>>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>>>> http://asipto.com/u/kat
>> -- 
>> Daniel-Constantin Mierla -
>> http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>> http://asipto.com/u/kat
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From tayeb at vobradio.org  Fri Aug 31 11:04:04 2012
From: tayeb at vobradio.org (Tayeb Meftah)
Date: Fri, 31 Aug 2012 13:04:04 +0400
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
Message-ID: <-2726116749635074150@unknownmsgid>

Hello
Sbc also ofer a propritary way of failover for itself
If node1 die node2 will replace it

Envoy? de mon iPhone

Le 31 ao?t 2012 ? 11:47, "Olle E. Johansson" <oej at edvina.net> a ?crit :

> Good discussion!
>
> In most, but not all, cases it's a political/business decision outside of the scope of the technichal specifications. A commercial SBC delivers a cloud of magic dust that makes some people feel better and more secure. I have audited several SBC installations that are totally insecure, where the local techies lack knowledge on how to operate it. Management people think the SBC is secure by design. I can't blame the vendors here - it's more correct to blame the decision process.
>
> You can always build an SBC with Kamailio plus a media server like sems, Asterisk or FreeSwitch in combination with IPtables. It requires experience and knowledge, but it is propably a better longterm to invest in local knowledge for security instead of relying on outside companies without building local knowledge...
>
> Another problem is the architecture issue. SIP took off as a PSTN replacement. The design of the SIP network and a traditional switch/PBX network is radically different, so SIP networks was built with non-SIP design, with heavy media servers (b2bua) controlling every call. This put a severe limitation in the service set and forced a big dependency into the design, that wasn't there in the original SIP design. In order for SIP to get anywhere beyond G711 calls we need to rip that apart.
>
> Trying to do that is hard, since the endpoints are more and more designed for this design. Normal phone services like music-on-hold and call parking are hard to deliver without support from the phones. That forces many design to move back to a PBX design.
>
> It takes time to change people's mindset. We need to prove that we can build better and interoperable solutions. I can't get SIP presence to work across vendors. That's bad for the industry, bad for us and hurts the trust in new designs and solutions. Which moves people back to the old trustworthy and reliant PBX vendor. And the SBC, because that's what gartner group and similar institutes tell them is the proper design.
>
> The battle isn't over. We need to prove our vision. Andreas and SIPwise is doing great work here, as is AG Projects. And, of course, the whole Kamailio community and excellent developer team.
>
> /O
>
> * The new Edvina SIP Masterclass - Stockholm Oct, Miami, FL Dec 2012
>  http://edvina.net/training/new-sip-masterclass/
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


From oej at edvina.net  Fri Aug 31 11:10:50 2012
From: oej at edvina.net (Olle E. Johansson)
Date: Fri, 31 Aug 2012 11:10:50 +0200
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <-2726116749635074150@unknownmsgid>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
	<-2726116749635074150@unknownmsgid>
Message-ID: <C3153457-E823-4247-AE8D-08898151837C@edvina.net>


31 aug 2012 kl. 11:04 skrev Tayeb Meftah <tayeb at vobradio.org>:

> Hello
> Sbc also ofer a propritary way of failover for itself
> If node1 die node2 will replace it

That's been done both with Asterisk and Kamailio (and propably SEMS) for a very long time.

/O
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From tayeb at vobradio.org  Fri Aug 31 11:15:00 2012
From: tayeb at vobradio.org (Tayeb Meftah)
Date: Fri, 31 Aug 2012 13:15:00 +0400
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <C3153457-E823-4247-AE8D-08898151837C@edvina.net>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
	<-2726116749635074150@unknownmsgid>
	<C3153457-E823-4247-AE8D-08898151837C@edvina.net>
Message-ID: <-8555328358638506920@unknownmsgid>

Yep
Freeswitch too
But go tel bt or any big carrier

Envoy? de mon iPhone

Le 31 ao?t 2012 ? 13:11, "Olle E. Johansson" <oej at edvina.net> a ?crit :

>
> 31 aug 2012 kl. 11:04 skrev Tayeb Meftah <tayeb at vobradio.org>:
>
>> Hello
>> Sbc also ofer a propritary way of failover for itself
>> If node1 die node2 will replace it
>
> That's been done both with Asterisk and Kamailio (and propably SEMS) for a very long time.
>
> /O
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


From agranig at sipwise.com  Fri Aug 31 11:52:42 2012
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 31 Aug 2012 11:52:42 +0200
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <C3153457-E823-4247-AE8D-08898151837C@edvina.net>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
	<-2726116749635074150@unknownmsgid>
	<C3153457-E823-4247-AE8D-08898151837C@edvina.net>
Message-ID: <5040896A.3090209@sipwise.com>

On 08/31/2012 11:10 AM, Olle E. Johansson wrote:
>> Sbc also ofer a propritary way of failover for itself
>> If node1 die node2 will replace it
> 
> That's been done both with Asterisk and Kamailio (and propably SEMS) for a very long time.

For SEMS it's a commercial module, same goes for the Sipwise mediaproxy
for seemless RTP stream fail-over. That's still where open source people
earn their money :)

Andreas

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From neill.wilkinson at btinternet.com  Fri Aug 31 12:12:27 2012
From: neill.wilkinson at btinternet.com (Neill Wilkinson)
Date: Fri, 31 Aug 2012 11:12:27 +0100
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <5040896A.3090209@sipwise.com>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
	<-2726116749635074150@unknownmsgid>
	<C3153457-E823-4247-AE8D-08898151837C@edvina.net>
	<5040896A.3090209@sipwise.com>
Message-ID: <CAJ-mRNNduJME-iF7sA5-5R4Kjs7tvjC4_hj9Rm2SEUxfWOPdKw@mail.gmail.com>

I think for a lot of customers its as much about commercial support from
vendors (basically the same story for all open source vs. commercial
solutions), as it is about technical choice.

SBCs are seen as a fix all perimeter control for VoIP. I think as we all
know there are many ways to solve a problem and architectural solutions for
VoIP "interconnect" can also be solved by placing the SBC infront of an
infrastructure (perimeter control) or as a border defence behind a
proxy/load-balancer.

In some instances its about simplicity. For example I deal with hosted
contact Centre environments built around commercial VoIP contact centre
solutions from Oracle and CosmoCom, for these it is simpler to place a
commercial SBC such as Acme, Sonus or even Cisco "CUBE" facing a carrier
SIP trunk interconnect. My customers have the assurance of vendor support
and I as a SME, I don't have put a significant amount of time and effort in
to supporting an Open Source solution. (Please don't flame me on this last
point). Whilst I would like to place Open Source products in this mix
(OpenSER/SER, SEMS, Asterisk, Freeswitch, etc) for some of my customers
they just don't want to go down that route.

Neill

Neill Wilkinson
Aeonvista Ltd


On 31 August 2012 10:52, Andreas Granig <agranig at sipwise.com> wrote:

> On 08/31/2012 11:10 AM, Olle E. Johansson wrote:
> >> Sbc also ofer a propritary way of failover for itself
> >> If node1 die node2 will replace it
> >
> > That's been done both with Asterisk and Kamailio (and propably SEMS) for
> a very long time.
>
> For SEMS it's a commercial module, same goes for the Sipwise mediaproxy
> for seemless RTP stream fail-over. That's still where open source people
> earn their money :)
>
> Andreas
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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From abalashov at evaristesys.com  Fri Aug 31 14:13:52 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 31 Aug 2012 08:13:52 -0400
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <CAJ-mRNNduJME-iF7sA5-5R4Kjs7tvjC4_hj9Rm2SEUxfWOPdKw@mail.gmail.com>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
	<-2726116749635074150@unknownmsgid>
	<C3153457-E823-4247-AE8D-08898151837C@edvina.net>
	<5040896A.3090209@sipwise.com>
	<CAJ-mRNNduJME-iF7sA5-5R4Kjs7tvjC4_hj9Rm2SEUxfWOPdKw@mail.gmail.com>
Message-ID: <5040AA80.7000905@evaristesys.com>

Commercial SBCs are quite featureful, but their feature set is extremely 
static, and often depends on vendor consulting and expertise to be 
approachable.

Nobody claimed the open-source solutions are simple to support or don't 
require expertise.  They're not free, either, as everyone well knows.

In my view, the real value of something like Kamailio - where it's not 
merely a money-saving measure but can achieve truly spectacular 
multiplier effects in technological leverage - is in its integration 
paths (MI and sercmd/binrpc), HTTP client/server, database connectors, 
integrated servers, etc.  The ability to develop new kinds of 
applications and services that rely on making disparate components talk 
to and interoperate with each other in novel ways, as well as to make 
better, more economical use of existing technology through those same 
kinds of facilities, is where the real power truly lies.  Some 
commercial products powered by proprietary stacks have APIs too, but 
they're light years behind the open-source world in this regard; they 
are far too bureaucratic and inflexible.

This is where our customers typically see massive payback of using 
something like Kamailio, aside from the any savings on licensing costs 
(both of the core network element and potential dependencies).

-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From abalashov at evaristesys.com  Fri Aug 31 14:18:24 2012
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 31 Aug 2012 08:18:24 -0400
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <5040AA80.7000905@evaristesys.com>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
	<-2726116749635074150@unknownmsgid>
	<C3153457-E823-4247-AE8D-08898151837C@edvina.net>
	<5040896A.3090209@sipwise.com>
	<CAJ-mRNNduJME-iF7sA5-5R4Kjs7tvjC4_hj9Rm2SEUxfWOPdKw@mail.gmail.com>
	<5040AA80.7000905@evaristesys.com>
Message-ID: <5040AB90.4040004@evaristesys.com>

Also, Kamailio's procedural route script language is a major asset of 
its particular methodology.

While it can sometimes be a pain for newbies trying to do relatively 
simple, canonical kind of things, the flexibility it affords in 
scripting SIP outcomes on a very granular level, and providing a 
powerful state machine within that programmatic framework, are both 
hugely important to "non-trivial" endeavours.

The feature set of something like an Acme Packet NetNet SD is 
essentially static; it does what it does, and not one bit more.  It may 
have a lot of capabilities, but those capabilities are all a matter of 
configuration, not of customisation or extension.  It is in something 
like Kamailio (or Asterisk or Freeswitch) that you can truly *extend* 
the functionality, allow it to interface with other open-source and 
proprietary components, build complex middleware and business layers, etc.

On 08/31/2012 08:13 AM, Alex Balashov wrote:

> Commercial SBCs are quite featureful, but their feature set is extremely
> static, and often depends on vendor consulting and expertise to be
> approachable.
>
> Nobody claimed the open-source solutions are simple to support or don't
> require expertise.  They're not free, either, as everyone well knows.
>
> In my view, the real value of something like Kamailio - where it's not
> merely a money-saving measure but can achieve truly spectacular
> multiplier effects in technological leverage - is in its integration
> paths (MI and sercmd/binrpc), HTTP client/server, database connectors,
> integrated servers, etc.  The ability to develop new kinds of
> applications and services that rely on making disparate components talk
> to and interoperate with each other in novel ways, as well as to make
> better, more economical use of existing technology through those same
> kinds of facilities, is where the real power truly lies.  Some
> commercial products powered by proprietary stacks have APIs too, but
> they're light years behind the open-source world in this regard; they
> are far too bureaucratic and inflexible.
>
> This is where our customers typically see massive payback of using
> something like Kamailio, aside from the any savings on licensing costs
> (both of the core network element and potential dependencies).
>


-- 
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


From fred at teamforrest.com  Fri Aug 31 14:25:41 2012
From: fred at teamforrest.com (Fred Posner)
Date: Fri, 31 Aug 2012 08:25:41 -0400
Subject: [SR-Users] [OT] the role of SBCs
In-Reply-To: <FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
References: <503F7AD9.8040302@gmail.com>
	<CAOCjumHqiA--KsKKZ1GWThjWVL4XGYhzub7aDKb4Zf=mf3O6nw@mail.gmail.com>,
	<503FB743.1000100@gmail.com>
	<8ABFC807392ABC49BCAD4465458B3C910416F094E9@EMV32-UKDY.domain1.systemhost.net>
	<503FE91A.6010503@sipwise.com>
	<FF96FCC7-8208-4AAE-82EB-326C04F3EB01@edvina.net>
Message-ID: <40A2F75D-EEEF-4E4B-A12E-7CA20BC4F75B@teamforrest.com>

When I first got into VoIP, my knowledge was less than stellar. The main decision make and I had believed that if we hired quality (and not cheap) SME we would be given great information and the money spent would pay for itself. We ended up working with a broadsoft system and an acme packet sbc. We were really sold that this would be the creme de la creme-- no nat issues, failover media, security, stability.

Crap. Problems galore, especially with residential NAT users. Despite having 2 acme's in a failover, an outage from the main isp resulted in a crippling thundering herd when connection was restored. Immediately, and know with some decent knowledge, I started working with (at the time) openser.

We deployed it within 2 weeks. There was no feature lost. In fact, we had only gains. All the NAT problems suddenly went away. We purposely tried to kill the openser with a thundering herd. Couldn't do it. There was a learning curve, but when is there not a learning curve?

Honestly, at that time... the savings (which were incredible) wasn't an issue. If it were more than an acme, we would have paid it. We needed something that worked, and the best product we could find was openser.

Since then, I've been a strong supporter. With the recent modifications (do we still consider anti_flood recent?), there's really no other choice for me. Yes, it takes programming, customization, and set-up. So does a commercial product. It's life.

When I first deployed kamailio, I didn't consider it an SBC. I considered it an SBC replacement. 

With best regards,

Fred
http://qxork.com

On Aug 31, 2012, at 3:47 AM, Olle E. Johansson wrote:

> In most, but not all, cases it's a political/business decision outside of the scope of the technichal specifications. A commercial SBC delivers a cloud of magic dust that makes some people feel better and more secure. I have audited several SBC installations that are totally insecure, where the local techies lack knowledge on how to operate it. Management people think the SBC is secure by design. I can't blame the vendors here - it's more correct to blame the decision process.



From jh at tutpro.com  Fri Aug 31 15:06:00 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 16:06:00 +0300
Subject: [SR-Users]  LCR (next_gw) question
In-Reply-To: <CAHViUgd_VUB_5YAv7_WyYMMnxkDBhrT5KXt1dZ8q-q7hQ-=5vQ@mail.gmail.com>
References: <CAHViUgd_VUB_5YAv7_WyYMMnxkDBhrT5KXt1dZ8q-q7hQ-=5vQ@mail.gmail.com>
Message-ID: <20544.46776.305106.197793@siika.tutpro.com>

Nord7 writes:

> Hello everyone. Configures the module LCR. I can't get him to take the next
> on the list of priorities gateway if the first does not work. Function
> "next_gw" capable of doing that? Did anyone realties that? If possible show
> example.

you first call load_gws and then next_gw for each gw to try.  load_gws
and first next_gw calls are typically in route block and subsequent
next_gw calls are in failure route block that you need to arm before
calling t_relay.

i don't know if there are any complete examples somewhere.

-- juha


From jh at tutpro.com  Fri Aug 31 15:13:32 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 16:13:32 +0300
Subject: [SR-Users] invite/ack avps/transactions
Message-ID: <20544.47228.978641.579287@siika.tutpro.com>

i tried to access an avp that was set during invite processing while
processing ack to 200 ok, but i guess the avp is already gone by that
time, because positive ack is its own transaction?

what i tried to do is to save somewhere invite's $du and then t_relay
ack to the same place without calling loose_route.

should i go for htable as storage or is there a better choice?

-- juha


From patrice.bodeven at orange.com  Fri Aug 31 16:09:27 2012
From: patrice.bodeven at orange.com (patrice.bodeven at orange.com)
Date: Fri, 31 Aug 2012 14:09:27 +0000
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <50407720.8070109@gmail.com>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
	<3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<50407720.8070109@gmail.com>
Message-ID: <27476_1346422169_5040C599_27476_7367_1_F51524208B2244439FE9395DA60B7EB6030BC9@PEXCVZYM14.corporate.adroot.infra.ftgroup>

Hello 

The return code is 2. 

The problem seems due to "nonce.c" 
if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) {
xxx
}

So I put 

	DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %s)\n", &b_nonce.n.md5_1[0]);
	DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 %s)\n", &b_nonce2.n.md5_1[0]);
	return 2;
}

The result into log is : 

Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: registrar [lookup.c:229]: '+33296053686' found in usrloc
Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [nonce.c:473]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce ????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [nonce.c:474]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 r?7B??| {?9R~R?FS?~Sb~@???#025n?| ???Zo#032#Y?P~@??#010PA#027#030P@??????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [api.c:147]: auth:pre_auth: Invalid nonce value received
Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [api.c:148]: auth:pre_auth: Invalid nonce value received (ret 2)

Cordialement/ Best regards 
Patrice B

-----Message d'origine-----
De?: Daniel-Constantin Mierla [mailto:miconda at gmail.com] 
Envoy??: vendredi 31 ao?t 2012 10:35
??: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
Cc?: BODEVEN Patrice RD-CORE
Objet?: Re: [SR-Users] Kamailio - Nonce validity between Register and Invite

Hello,

On 8/30/12 10:43 AM, patrice.bodeven at orange.com wrote:
> Hello,
>
> Sorry, but i am blocked on the nonce reuse between the Register and the INVITE even with the info provided in debug mode.
>   
> My understanding is Nonce should be valid between SIP methods until the nonce is valid in the time.
> But I don't know how the Nonce is stored and how Kamailio is looking for the nonce into the memory !
>
> I did a test on Register => nonce reuse is ok.
> I did a test on INVITE => nonce reuse of Register or on previous INVITE doesn't work !
>
> But based on the site indicated below and the SIPP (generate the INVITE with the Nonce of Register), the result is
> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>
> But there is no explanation why the nonce is invalid.
can you apply the following patch:

http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/auth/api.c;h=c79de5c9dd6cd3e176c05d836c7b0a4275f8a3d2;hp=2ee39a7a6cbcce741365dac97ae8a3db8c196b0c;hb=494b383edde7a2d193c220f3117506e4cc95932f;hpb=e4ecf49add0e62330e1db071106806e35e9b078a

It practically adds the return code in the log message, that should give 
better leads to why the nonce is considered invalid, allowing to track 
back quicker in the source code.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat


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From miconda at gmail.com  Fri Aug 31 17:04:12 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 17:04:12 +0200
Subject: [SR-Users] invite/ack avps/transactions
In-Reply-To: <20544.47228.978641.579287@siika.tutpro.com>
References: <20544.47228.978641.579287@siika.tutpro.com>
Message-ID: <5040D26C.9000205@gmail.com>

Hello,

On 8/31/12 3:13 PM, Juha Heinanen wrote:
> i tried to access an avp that was set during invite processing while
> processing ack to 200 ok, but i guess the avp is already gone by that
> time, because positive ack is its own transaction?
>
> what i tried to do is to save somewhere invite's $du and then t_relay
> ack to the same place without calling loose_route.
>
> should i go for htable as storage or is there a better choice?
htable is the solution, ACK for 200ok being a separate transaction.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Fri Aug 31 17:08:35 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 17:08:35 +0200
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <27476_1346422169_5040C599_27476_7367_1_F51524208B2244439FE9395DA60B7EB6030BC9@PEXCVZYM14.corporate.adroot.infra.ftgroup>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
	<3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<50407720.8070109@gmail.com>
	<27476_1346422169_5040C599_27476_7367_1_F51524208B2244439FE9395DA60B7EB6030BC9@PEXCVZYM14.corporate.adroot.infra.ftgroup>
Message-ID: <5040D373.4050604@gmail.com>

Hello,

I will look over the code, I just wanted to point that it is safer to 
print by size in this case, otherwise it can get coredump, even if the 
printed characters are not ascii, like:

DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n", &b_nonce.n.md5_1[0], 16);


Otherwise DBG(..) goes on until finds '\0', which can be in a memory 
zone used by another process.

Cheers,
Daniel

On 8/31/12 4:09 PM, patrice.bodeven at orange.com wrote:
> Hello
>
> The return code is 2.
>
> The problem seems due to "nonce.c"
> if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) {
> xxx
> }
>
> So I put
>
> 	DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %s)\n", &b_nonce.n.md5_1[0]);
> 	DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 %s)\n", &b_nonce2.n.md5_1[0]);
> 	return 2;
> }
>
> The result into log is :
>
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: registrar [lookup.c:229]: '+33296053686' found in usrloc
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [nonce.c:473]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce ????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [nonce.c:474]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 r?7B??| {?9R~R?FS?~Sb~@???#025n?| ???Zo#032#Y?P~@??#010PA#027#030P@??????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [api.c:147]: auth:pre_auth: Invalid nonce value received
> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: auth [api.c:148]: auth:pre_auth: Invalid nonce value received (ret 2)
>
> Cordialement/ Best regards
> Patrice B
>
> -----Message d'origine-----
> De : Daniel-Constantin Mierla [mailto:miconda at gmail.com]
> Envoy? : vendredi 31 ao?t 2012 10:35
> ? : SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
> Cc : BODEVEN Patrice RD-CORE
> Objet : Re: [SR-Users] Kamailio - Nonce validity between Register and Invite
>
> Hello,
>
> On 8/30/12 10:43 AM, patrice.bodeven at orange.com wrote:
>> Hello,
>>
>> Sorry, but i am blocked on the nonce reuse between the Register and the INVITE even with the info provided in debug mode.
>>    
>> My understanding is Nonce should be valid between SIP methods until the nonce is valid in the time.
>> But I don't know how the Nonce is stored and how Kamailio is looking for the nonce into the memory !
>>
>> I did a test on Register => nonce reuse is ok.
>> I did a test on INVITE => nonce reuse of Register or on previous INVITE doesn't work !
>>
>> But based on the site indicated below and the SIPP (generate the INVITE with the Nonce of Register), the result is
>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>>
>> But there is no explanation why the nonce is invalid.
> can you apply the following patch:
>
> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/auth/api.c;h=c79de5c9dd6cd3e176c05d836c7b0a4275f8a3d2;hp=2ee39a7a6cbcce741365dac97ae8a3db8c196b0c;hb=494b383edde7a2d193c220f3117506e4cc95932f;hpb=e4ecf49add0e62330e1db071106806e35e9b078a
>
> It practically adds the return code in the log message, that should give
> better leads to why the nonce is considered invalid, allowing to track
> back quicker in the source code.
>
> Cheers,
> Daniel
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Fri Aug 31 17:10:35 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 17:10:35 +0200
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <5040D373.4050604@gmail.com>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
	<3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<50407720.8070109@gmail.com>
	<27476_1346422169_5040C599_27476_7367_1_F51524208B2244439FE9395DA60B7EB6030BC9@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5040D373.4050604@gmail.com>
Message-ID: <5040D3EB.50100@gmail.com>

Hello,


On 8/31/12 5:08 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I will look over the code, I just wanted to point that it is safer to 
> print by size in this case, otherwise it can get coredump, even if the 
> printed characters are not ascii, like:
>
> DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n", 
> &b_nonce.n.md5_1[0], 16);

actually the size has to be first, like:

DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n", 16, 
&b_nonce.n.md5_1[0]);

Cheers,
Daniel

>
>
> Otherwise DBG(..) goes on until finds '\0', which can be in a memory 
> zone used by another process.
>
> Cheers,
> Daniel
>
> On 8/31/12 4:09 PM, patrice.bodeven at orange.com wrote:
>> Hello
>>
>> The return code is 2.
>>
>> The problem seems due to "nonce.c"
>> if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) {
>> xxx
>> }
>>
>> So I put
>>
>>     DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %s)\n", 
>> &b_nonce.n.md5_1[0]);
>>     DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 %s)\n", 
>> &b_nonce2.n.md5_1[0]);
>>     return 2;
>> }
>>
>> The result into log is :
>>
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> registrar [lookup.c:229]: '+33296053686' found in usrloc
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [nonce.c:473]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce 
>> ????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [nonce.c:474]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 
>> r?7B??| {?9R~R?FS?~Sb~@???#025n?| 
>> ???Zo#032#Y?P~@??#010PA#027#030P@??????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| 
>> ???Zo#032#Y?P~@??#010)
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [api.c:148]: auth:pre_auth: Invalid nonce value received (ret 2)
>>
>> Cordialement/ Best regards
>> Patrice B
>>
>> -----Message d'origine-----
>> De : Daniel-Constantin Mierla [mailto:miconda at gmail.com]
>> Envoy? : vendredi 31 ao?t 2012 10:35
>> ? : SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - 
>> Users Mailing List
>> Cc : BODEVEN Patrice RD-CORE
>> Objet : Re: [SR-Users] Kamailio - Nonce validity between Register and 
>> Invite
>>
>> Hello,
>>
>> On 8/30/12 10:43 AM, patrice.bodeven at orange.com wrote:
>>> Hello,
>>>
>>> Sorry, but i am blocked on the nonce reuse between the Register and 
>>> the INVITE even with the info provided in debug mode.
>>>    My understanding is Nonce should be valid between SIP methods 
>>> until the nonce is valid in the time.
>>> But I don't know how the Nonce is stored and how Kamailio is looking 
>>> for the nonce into the memory !
>>>
>>> I did a test on Register => nonce reuse is ok.
>>> I did a test on INVITE => nonce reuse of Register or on previous 
>>> INVITE doesn't work !
>>>
>>> But based on the site indicated below and the SIPP (generate the 
>>> INVITE with the Nonce of Register), the result is
>>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>>>
>>> But there is no explanation why the nonce is invalid.
>> can you apply the following patch:
>>
>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/auth/api.c;h=c79de5c9dd6cd3e176c05d836c7b0a4275f8a3d2;hp=2ee39a7a6cbcce741365dac97ae8a3db8c196b0c;hb=494b383edde7a2d193c220f3117506e4cc95932f;hpb=e4ecf49add0e62330e1db071106806e35e9b078a 
>>
>>
>> It practically adds the return code in the log message, that should give
>> better leads to why the nonce is considered invalid, allowing to track
>> back quicker in the source code.
>>
>> Cheers,
>> Daniel
>>
>


-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From jh at tutpro.com  Fri Aug 31 18:15:13 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 19:15:13 +0300
Subject: [SR-Users] weird dp_translate result
Message-ID: <20544.58129.485514.359848@siika.tutpro.com>

i have this piece of config code:

modparam("dialplan", "attrs_pvar", "$var(plan)")
...
    xlog("L_INFO", "Translating <$var(calling_number)> based on dpid 4\n");

    if (dp_translate("4", "$var(calling_number)")) {
	xlog("L_INFO", "Found attrs <$var(plan)>\n");
	...

dpid 4 has three entries:

109	4	10	1	^\+1	0		na
107	4	10	1	^\+358	0		fi
108	4	10	1	^\+46	0		se

thus in each entry attrs field has some two letter value.

when the above piece of code is executed, i get to syslog

Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Translating <+35892345670> based on dpid 4
Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Found attrs <0>

if i restart the proxy, attrs value may change to some other bogus
value.

any ideas?

-- juha




From patrice.bodeven at orange.com  Fri Aug 31 18:18:52 2012
From: patrice.bodeven at orange.com (patrice.bodeven at orange.com)
Date: Fri, 31 Aug 2012 16:18:52 +0000
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <5040D3EB.50100@gmail.com>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
	<3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<50407720.8070109@gmail.com>
	<27476_1346422169_5040C599_27476_7367_1_F51524208B2244439FE9395DA60B7EB6030BC9@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5040D373.4050604@gmail.com> <5040D3EB.50100@gmail.com>
Message-ID: <9094_1346429933_5040E3ED_9094_3903_1_F51524208B2244439FE9395DA60B7EB6030C3F@PEXCVZYM14.corporate.adroot.infra.ftgroup>

Hello, 

I find my error finally in my configuration.

The initial setting was only  

modparam("auth", "auth_checks_register", 6) # callid and From TAG

I forgot to set also the other auth_checks on no_dlg and in_dlg. 

modparam("auth", "auth_checks_no_dlg", 6) # callid and From TAG
modparam("auth", "auth_checks_in_dlg", 6) # callid and From TAG

Thank you for your support, sorry for the disturbing. 

Cordialement/ Best regards 
Patrice B


-----Message d'origine-----
De?: Daniel-Constantin Mierla [mailto:miconda at gmail.com] 
Envoy??: vendredi 31 ao?t 2012 17:11
??: miconda at gmail.com; SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
Cc?: BODEVEN Patrice RD-CORE
Objet?: Re: [SR-Users] Kamailio - Nonce validity between Register and Invite

Hello,


On 8/31/12 5:08 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I will look over the code, I just wanted to point that it is safer to 
> print by size in this case, otherwise it can get coredump, even if the 
> printed characters are not ascii, like:
>
> DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n", 
> &b_nonce.n.md5_1[0], 16);

actually the size has to be first, like:

DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n", 16, 
&b_nonce.n.md5_1[0]);

Cheers,
Daniel

>
>
> Otherwise DBG(..) goes on until finds '\0', which can be in a memory 
> zone used by another process.
>
> Cheers,
> Daniel
>
> On 8/31/12 4:09 PM, patrice.bodeven at orange.com wrote:
>> Hello
>>
>> The return code is 2.
>>
>> The problem seems due to "nonce.c"
>> if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) {
>> xxx
>> }
>>
>> So I put
>>
>>     DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %s)\n", 
>> &b_nonce.n.md5_1[0]);
>>     DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 %s)\n", 
>> &b_nonce2.n.md5_1[0]);
>>     return 2;
>> }
>>
>> The result into log is :
>>
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> registrar [lookup.c:229]: '+33296053686' found in usrloc
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [nonce.c:473]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce 
>> ????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [nonce.c:474]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 
>> r?7B??| {?9R~R?FS?~Sb~@???#025n?| 
>> ???Zo#032#Y?P~@??#010PA#027#030P@??????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| 
>> ???Zo#032#Y?P~@??#010)
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG: 
>> auth [api.c:148]: auth:pre_auth: Invalid nonce value received (ret 2)
>>
>> Cordialement/ Best regards
>> Patrice B
>>
>> -----Message d'origine-----
>> De : Daniel-Constantin Mierla [mailto:miconda at gmail.com]
>> Envoy? : vendredi 31 ao?t 2012 10:35
>> ? : SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - 
>> Users Mailing List
>> Cc : BODEVEN Patrice RD-CORE
>> Objet : Re: [SR-Users] Kamailio - Nonce validity between Register and 
>> Invite
>>
>> Hello,
>>
>> On 8/30/12 10:43 AM, patrice.bodeven at orange.com wrote:
>>> Hello,
>>>
>>> Sorry, but i am blocked on the nonce reuse between the Register and 
>>> the INVITE even with the info provided in debug mode.
>>>    My understanding is Nonce should be valid between SIP methods 
>>> until the nonce is valid in the time.
>>> But I don't know how the Nonce is stored and how Kamailio is looking 
>>> for the nonce into the memory !
>>>
>>> I did a test on Register => nonce reuse is ok.
>>> I did a test on INVITE => nonce reuse of Register or on previous 
>>> INVITE doesn't work !
>>>
>>> But based on the site indicated below and the SIPP (generate the 
>>> INVITE with the Nonce of Register), the result is
>>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>>>
>>> But there is no explanation why the nonce is invalid.
>> can you apply the following patch:
>>
>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/auth/api.c;h=c79de5c9dd6cd3e176c05d836c7b0a4275f8a3d2;hp=2ee39a7a6cbcce741365dac97ae8a3db8c196b0c;hb=494b383edde7a2d193c220f3117506e4cc95932f;hpb=e4ecf49add0e62330e1db071106806e35e9b078a 
>>
>>
>> It practically adds the return code in the log message, that should give
>> better leads to why the nonce is considered invalid, allowing to track
>> back quicker in the source code.
>>
>> Cheers,
>> Daniel
>>
>


-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat


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From miconda at gmail.com  Fri Aug 31 18:43:06 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 18:43:06 +0200
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <20544.58129.485514.359848@siika.tutpro.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
Message-ID: <6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>

Hello,

What version are you using? Is it happening to get the right value from time to time? 

If you set higher debug mode, is it printing the message that a rule was matched?

Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com

On 31 Aug 2012, at 18:15, Juha Heinanen <jh at tutpro.com> wrote:

> i have this piece of config code:
> 
> modparam("dialplan", "attrs_pvar", "$var(plan)")
> ...
>    xlog("L_INFO", "Translating <$var(calling_number)> based on dpid 4\n");
> 
>    if (dp_translate("4", "$var(calling_number)")) {
>    xlog("L_INFO", "Found attrs <$var(plan)>\n");
>    ...
> 
> dpid 4 has three entries:
> 
> 109    4    10    1    ^\+1    0        na
> 107    4    10    1    ^\+358    0        fi
> 108    4    10    1    ^\+46    0        se
> 
> thus in each entry attrs field has some two letter value.
> 
> when the above piece of code is executed, i get to syslog
> 
> Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Translating <+35892345670> based on dpid 4
> Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Found attrs <0>
> 
> if i restart the proxy, attrs value may change to some other bogus
> value.
> 
> any ideas?
> 
> -- juha
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


From spencer at 5ninesolutions.com  Fri Aug 31 18:49:51 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 31 Aug 2012 09:49:51 -0700
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <50407B8C.6090503@gmail.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
	<6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
	<503FB817.4060009@gmail.com>
	<B03A90C3-6658-4F80-94BE-7EBB7D39287E@5ninesolutions.com>
	<50407B8C.6090503@gmail.com>
Message-ID: <76e2de03d32bc5ec4e07e04bfebb4bdf@mail.5ninesolutions.com>

  

Yes, 

The request (re-INVITE): 

Aug 30 22:38:59 sip
/usr/sbin/kamailio[25778]: ERROR: *** cfgtrace:
c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
Aug 30
22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command
"25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 184.170.249.3 32122
3ae1Dvgr5vmeg;1 199857477;1"
Aug 30 22:38:59 sip rtpproxy[25671]:
INFO:handle_command: adding strong flag to existing session,
new=1/0/0
Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command:
lookup on ports 55324/46010, session timer restarted
Aug 30 22:38:59 sip
rtpproxy[25671]: DBUG:doreply: sending reply "25778_11 46010
184.170.249.8#012" 

The reply: 

Aug 30 22:38:59 sip
/usr/sbin/kamailio[25778]: ERROR: *** cfgtrace:
c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
Aug 30
22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command
"25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 71.104.248.48 6016
3ae1Dvgr5vmeg;1"
Aug 30 22:38:59 sip rtpproxy[25671]:
INFO:handle_command: lookup request failed: session
1952045641-6076-15 at BA.FJ.B.CFC, tags 3ae1Dvgr5vmeg;1/NONE not found
Aug
30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply "25778_12 0
184.170.249.8#012"
Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR:
rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy 

I
was able to get it working correctly by reworking the config like the
3.1 branch by using rtpproxy_offer instead of force_rtp_proxy. When I
attempted to use rtpproxy_answer in the reply route, I was getting the
same lookup request failed error from rtpproxy. In the request and
reply, the tags change. Could this be the reason that the session lookup
is failing? If I use rtpproxy_offer in both the request and reply,
everything works correctly. Is there any consequence to doing this?


Thanks, 

Spencer 

On 31.08.2012 01:53, Daniel-Constantin Mierla
wrote: 

> Hello,
> 
> On 8/31/12 3:41 AM, Spencer Thomason wrote:
> 
>>
Hi Daniel, I can confirm that rtpproxy_manage is called. See:
http://pastebin.com/ZVXjK9ry [1] I'm seeing ERROR: rtpproxy
[rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy in the logs
when processing the 200OK in the re-INVITE. I've included a debug level
log from rtpproxy in the log as well.
> 
> this can happen because the
rtpproxy was not engaged for the request, 
> but only for the reply.
>

> As you say, the logs are for the 200OK, what about the ones for
request, 
> is rtpproxy called there?
> 
> Cheers,
> Daniel
> 
>> When
handling the re-INVITE there is: * Aug 30 22:38:59 sip
/usr/sbin/kamailio[25778]: ERROR: *** cfgtrace:
c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage * Aug 30
22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command
"25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC [14] 184.170.249.3 32122
3ae1Dvgr5vmeg;1 199857477;1" * Aug 30 22:38:59 sip rtpproxy[25671]:
INFO:handle_command: adding strong flag to existing session, new=1/0/0 *
Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup on
ports 55324/46010, session timer restarted * Aug 30 22:38:59 sip
rtpproxy[25671]: DBUG:doreply: sending reply "25778_11 46010
184.170.249.8#012" but the 200OK: * Aug 30 22:38:59 sip
/usr/sbin/kamailio[25778]: ERROR: *** cfgtrace:
c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage * Aug 30
22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command
"25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC [15] 71.104.248.48 6016
3ae1Dvgr5vmeg;1" * Aug 30 22:38:59 sip rtpproxy[25671]:
INFO:handle_command: lookup request failed: session
1952045641-6076-15 at BA.FJ.B.CFC [16], tags 3ae1Dvgr5vmeg;1/NONE not found
* Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply
"25778_12 0 184.170.249.8#012" * Aug 30 22:38:59 sip
/usr/sbin/kamailio[25778]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect
port 0 in reply from rtp proxy I'm not familiar with the rtpproxy
commands to know why it cannot locate the session. Thanks for your
assistance, Spencer On Aug 30, 2012, at 11:59 AM, Daniel-Constantin
Mierla wrote: 
>> 
>>> Hello, I could not spot by quick eye checking
what could happen there, the best is to use the debugger module with
cfg_trace parameter set and check the execution trace to see what
actions of the configuration file are executed and be sure the rtpproxy
is called or not. You can post the execution trace here if you need
further help with it. Cheers, Daniel On 8/30/12 7:40 PM, Spencer
Thomason wrote: 
>>> 
>>>> Hi Daniel, Thanks for your help with this.
Here is a trace: http://pastebin.com/pXeFbwBz [8] I see the nat=yes
parameter added to the Route header. I've posted the script here:
http://pastebin.com/2qwHYHvj [9]Forgive my ignorance, I can't tell what
I'm doing wrong. Thanks! Spencer On Aug 30, 2012, at 12:51 AM,
Daniel-Constantin Mierla wrote: 
>>>> 
>>>>> Hello, if your config it is
based on the default one, the Route header for within dialog requests is
marked by a parameter, nat=yes, that is used to decide whether to do
rtpproxy or not. So, if you have a custom config, check the default one
for the nat traversal handling part. Cheers, Daniel On 8/30/12 12:39 AM,
Spencer Thomason wrote: 
>>>>> 
>>>>>> Hello, I'm using Kamailio 3.2.4
for NAT traversal using rtpproxy_manage() in a largely stock script.
Everything works great until the far end (on a public ip) sends a t.38
re-INVITE. The 200OK from the NATed UAC then doesn't trigger rtpproxy
and the private IP in the sdp causes the fax to fail. Any help handling
these re-INVITEs would be greatly appreciated. I'm happy to post traces
if that helps describe the situation. The network topology looks like
this: NATed UAC -> Kamailio on a public IP -> UAS on a public IP Thanks
in advance, Spencer Connected by DROID on Verizon Wireless
_______________________________________________ SIP Express Router (SER)
and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org [2]
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users [3]
>>>>>
-- Daniel-Constantin Mierla - http://www.asipto.com [4]
http://twitter.com/# [5]!/miconda - http://www.linkedin.com/in/miconda
[6] Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
http://asipto.com/u/kat [7]
>>> -- Daniel-Constantin Mierla -
http://www.asipto.com [10] http://twitter.com/# [11]!/miconda -
http://www.linkedin.com/in/miconda [12] Kamailio Advanced Training,
Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat [13]
>>
_______________________________________________ SIP Express Router (SER)
and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org [17]
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users [18]



Links:
------
[1] http://pastebin.com/ZVXjK9ry
[2]
mailto:sr-users at lists.sip-router.org
[3]
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[4]
http://www.asipto.com
[5] http://twitter.com/#
[6]
http://www.linkedin.com/in/miconda
[7] http://asipto.com/u/kat
[8]
http://pastebin.com/pXeFbwBz
[9] http://pastebin.com/2qwHYHvj
[10]
http://www.asipto.com
[11] http://twitter.com/#
[12]
http://www.linkedin.com/in/miconda
[13] http://asipto.com/u/kat
[14]
mailto:1952045641-6076-15 at BA.FJ.B.CFC
[15]
mailto:1952045641-6076-15 at BA.FJ.B.CFC
[16]
mailto:1952045641-6076-15 at BA.FJ.B.CFC
[17]
mailto:sr-users at lists.sip-router.org
[18]
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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From jh at tutpro.com  Fri Aug 31 19:25:47 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 20:25:47 +0300
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
	<6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
Message-ID: <20544.62363.353350.416390@siika.tutpro.com>

Daniel-Constantin Mierla writes:

> What version are you using? Is it happening to get the right value
> from time to time?

this is with a few days old trunk:

version: sip-proxy 3.4.0-dev3 (x86_64/linux) 

i have seen it sometimes to set attrs pvar correctly, but most times
not.

> If you set higher debug mode, is it printing the message that a rule
> was matched?

see below.

-- juha

Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: INFO: Translating <+35892345670> based on dpid 4
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:240]: integer value
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:261]: searching 15 
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:348]: input is +35892345670
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:341]: regex operator testing
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:341]: regex operator testing
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:341]: regex operator testing
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:386]: found a matching rule 0x7f7d72275fa8: pr 10, match_exp ^\+358
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:393]: the rule's attrs are fi
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:404]: the copied attributes are: fi
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:151]: null replacement
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:357]: input +35892345670 with dpid 4 => output 
Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: INFO: Found attrs <<null>>


From miconda at gmail.com  Fri Aug 31 20:18:12 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 20:18:12 +0200
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <20544.62363.353350.416390@siika.tutpro.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
	<6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
	<20544.62363.353350.416390@siika.tutpro.com>
Message-ID: <5040FFE4.6040608@gmail.com>

I checked the commit log and there was no change in the module. Failure 
to set the attrs variable is printing a error message, so looks like the 
variable is set.

Can you set $var(plan) to a static string value before dp_translate() 
just to see if the value is updated in the function?

Cheers,
Daniel

On 8/31/12 7:25 PM, Juha Heinanen wrote:
> Daniel-Constantin Mierla writes:
>
>> What version are you using? Is it happening to get the right value
>> from time to time?
> this is with a few days old trunk:
>
> version: sip-proxy 3.4.0-dev3 (x86_64/linux)
>
> i have seen it sometimes to set attrs pvar correctly, but most times
> not.
>
>> If you set higher debug mode, is it printing the message that a rule
>> was matched?
> see below.
>
> -- juha
>
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: INFO: Translating <+35892345670> based on dpid 4
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:240]: integer value
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:261]: searching 15
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:348]: input is +35892345670
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:341]: regex operator testing
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:341]: regex operator testing
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:341]: regex operator testing
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:386]: found a matching rule 0x7f7d72275fa8: pr 10, match_exp ^\+358
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:393]: the rule's attrs are fi
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:404]: the copied attributes are: fi
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dp_repl.c:151]: null replacement
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: DEBUG: dialplan [dialplan.c:357]: input +35892345670 with dpid 4 => output
> Aug 31 20:21:56 wheezy2 /usr/sbin/sip-proxy[7560]: INFO: Found attrs <<null>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From miconda at gmail.com  Fri Aug 31 20:33:58 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 20:33:58 +0200
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <76e2de03d32bc5ec4e07e04bfebb4bdf@mail.5ninesolutions.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
	<6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
	<503FB817.4060009@gmail.com>
	<B03A90C3-6658-4F80-94BE-7EBB7D39287E@5ninesolutions.com>
	<50407B8C.6090503@gmail.com>
	<76e2de03d32bc5ec4e07e04bfebb4bdf@mail.5ninesolutions.com>
Message-ID: <50410396.2050805@gmail.com>

Hello,

the command to rtpproxy for the reply seem to miss the to-tag, can you 
grab the ngrep trace for such call and the logs for processing it? 
Having the logs from a different call than the ngrep trace you posted on 
pastebin is not helping much.

Cheers,
Daniel


On 8/31/12 6:49 PM, Spencer Thomason wrote:
>
> Yes,
>
> The request (re-INVITE):
>
> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received 
> command "25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 184.170.249.3 32122 
> 3ae1Dvgr5vmeg;1 199857477;1"
> Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: adding 
> strong flag to existing session, new=1/0/0
> Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup on 
> ports 55324/46010, session timer restarted
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply 
> "25778_11 46010 184.170.249.8#012"
>
> The reply:
>
> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received 
> command "25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 71.104.248.48 6016 
> 3ae1Dvgr5vmeg;1"
> Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup 
> request failed: session 1952045641-6076-15 at BA.FJ.B.CFC, tags 
> 3ae1Dvgr5vmeg;1/NONE not found
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply 
> "25778_12 0 184.170.249.8#012"
> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: rtpproxy 
> [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy
>
> I was able to get it working correctly by reworking the config like 
> the 3.1 branch by using rtpproxy_offer instead of force_rtp_proxy. 
>  When I attempted to use rtpproxy_answer in the reply route, I was 
> getting the same lookup request failed error from rtpproxy.  In the 
> request and reply, the tags change.  Could this be the reason that the 
> session lookup is failing?  If I use rtpproxy_offer in both the 
> request and reply, everything works correctly.  Is there any 
> consequence to doing this?
>
> Thanks,
>
> Spencer
>
> On 31.08.2012 01:53, Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> On 8/31/12 3:41 AM, Spencer Thomason wrote:
>>> Hi Daniel, I can confirm that rtpproxy_manage is called. See: 
>>> http://pastebin.com/ZVXjK9ry I'm seeing ERROR: rtpproxy 
>>> [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy in the 
>>> logs when processing the 200OK in the re-INVITE. I've included a 
>>> debug level log from rtpproxy in the log as well.
>> this can happen because the rtpproxy was not engaged for the request,
>> but only for the reply.
>>
>> As you say, the logs are for the 200OK, what about the ones for request,
>> is rtpproxy called there?
>>
>> Cheers,
>> Daniel
>>> When handling the re-INVITE there is: ? Aug 30 22:38:59 sip 
>>> /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
>>> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage ? Aug 30 
>>> 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command 
>>> "25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 
>>> <mailto:1952045641-6076-15 at BA.FJ.B.CFC> 184.170.249.3 32122 
>>> 3ae1Dvgr5vmeg;1 199857477;1" ? Aug 30 22:38:59 sip rtpproxy[25671]: 
>>> INFO:handle_command: adding strong flag to existing session, 
>>> new=1/0/0 ? Aug 30 22:38:59 sip rtpproxy[25671]: 
>>> INFO:handle_command: lookup on ports 55324/46010, session timer 
>>> restarted ? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: 
>>> sending reply "25778_11 46010 184.170.249.8#012" but the 200OK: ? 
>>> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
>>> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage ? Aug 30 
>>> 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command 
>>> "25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 
>>> <mailto:1952045641-6076-15 at BA.FJ.B.CFC> 71.104.248.48 6016 
>>> 3ae1Dvgr5vmeg;1" ? Aug 30 22:38:59 sip rtpproxy[25671]: 
>>> INFO:handle_command: lookup request failed: session 
>>> 1952045641-6076-15 at BA.FJ.B.CFC 
>>> <mailto:1952045641-6076-15 at BA.FJ.B.CFC>, tags 3ae1Dvgr5vmeg;1/NONE 
>>> not found ? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: 
>>> sending reply "25778_12 0 184.170.249.8#012" ? Aug 30 22:38:59 sip 
>>> /usr/sbin/kamailio[25778]: ERROR: rtpproxy [rtpproxy.c:2260]: 
>>> incorrect port 0 in reply from rtp proxy I'm not familiar with the 
>>> rtpproxy commands to know why it cannot locate the session. Thanks 
>>> for your assistance, Spencer On Aug 30, 2012, at 11:59 AM, 
>>> Daniel-Constantin Mierla wrote:
>>>> Hello, I could not spot by quick eye checking what could happen 
>>>> there, the best is to use the debugger module with cfg_trace 
>>>> parameter set and check the execution trace to see what actions of 
>>>> the configuration file are executed and be sure the rtpproxy is 
>>>> called or not. You can post the execution trace here if you need 
>>>> further help with it. Cheers, Daniel On 8/30/12 7:40 PM, Spencer 
>>>> Thomason wrote:
>>>>> Hi Daniel, Thanks for your help with this. Here is a trace: 
>>>>> http://pastebin.com/pXeFbwBz I see the nat=yes parameter added to 
>>>>> the Route header. I've posted the script here: 
>>>>> http://pastebin.com/2qwHYHvjForgive my ignorance, I can't tell 
>>>>> what I'm doing wrong. Thanks! Spencer On Aug 30, 2012, at 12:51 
>>>>> AM, Daniel-Constantin Mierla wrote:
>>>>>> Hello, if your config it is based on the default one, the Route 
>>>>>> header for within dialog requests is marked by a parameter, 
>>>>>> nat=yes, that is used to decide whether to do rtpproxy or not. 
>>>>>> So, if you have a custom config, check the default one for the 
>>>>>> nat traversal handling part. Cheers, Daniel On 8/30/12 12:39 AM, 
>>>>>> Spencer Thomason wrote:
>>>>>>> Hello, I'm using Kamailio 3.2.4 for NAT traversal using 
>>>>>>> rtpproxy_manage() in a largely stock script. Everything works 
>>>>>>> great until the far end (on a public ip) sends a t.38 re-INVITE. 
>>>>>>> The 200OK from the NATed UAC then doesn't trigger rtpproxy and 
>>>>>>> the private IP in the sdp causes the fax to fail. Any help 
>>>>>>> handling these re-INVITEs would be greatly appreciated. I'm 
>>>>>>> happy to post traces if that helps describe the situation. The 
>>>>>>> network topology looks like this: NATed UAC -> Kamailio on a 
>>>>>>> public IP -> UAS on a public IP Thanks in advance, Spencer 
>>>>>>> Connected by DROID on Verizon Wireless 
>>>>>>> _______________________________________________ SIP Express 
>>>>>>> Router (SER) and Kamailio (OpenSER) - sr-users mailing list 
>>>>>>> sr-users at lists.sip-router.org 
>>>>>>> <mailto:sr-users at lists.sip-router.org> 
>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>> -- Daniel-Constantin Mierla - http://www.asipto.com 
>>>>>> http://twitter.com/#!/miconda - 
>>>>>> http://www.linkedin.com/in/miconda Kamailio Advanced Training, 
>>>>>> Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>>> -- Daniel-Constantin Mierla - http://www.asipto.com 
>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda 
>>>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - 
>>>> http://asipto.com/u/kat
>>> _______________________________________________ SIP Express Router 
>>> (SER) and Kamailio (OpenSER) - sr-users mailing list 
>>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org> 
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Fri Aug 31 20:38:48 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 20:38:48 +0200
Subject: [SR-Users] Kamailio - Nonce validity between Register and Invite
In-Reply-To: <9094_1346429933_5040E3ED_9094_3903_1_F51524208B2244439FE9395DA60B7EB6030C3F@PEXCVZYM14.corporate.adroot.infra.ftgroup>
References: <669_1345625487_50349D8F_669_7953_1_F51524208B2244439FE9395DA60B7EB602FBA6@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5035C662.9000203@pernau.at>
	<3396_1346316195_503F27A3_3396_450_1_F51524208B2244439FE9395DA60B7EB603088D@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<50407720.8070109@gmail.com>
	<27476_1346422169_5040C599_27476_7367_1_F51524208B2244439FE9395DA60B7EB6030BC9@PEXCVZYM14.corporate.adroot.infra.ftgroup>
	<5040D373.4050604@gmail.com> <5040D3EB.50100@gmail.com>
	<9094_1346429933_5040E3ED_9094_3903_1_F51524208B2244439FE9395DA60B7EB6030C3F@PEXCVZYM14.corporate.adroot.infra.ftgroup>
Message-ID: <504104B8.1080400@gmail.com>

Hello,

great that you sorted out and thanks for posting the results so others 
can learn from them when searching the archive.

Cheers,
Daniel

On 8/31/12 6:18 PM, patrice.bodeven at orange.com wrote:
> Hello,
>
> I find my error finally in my configuration.
>
> The initial setting was only
>
> modparam("auth", "auth_checks_register", 6) # callid and From TAG
>
> I forgot to set also the other auth_checks on no_dlg and in_dlg.
>
> modparam("auth", "auth_checks_no_dlg", 6) # callid and From TAG
> modparam("auth", "auth_checks_in_dlg", 6) # callid and From TAG
>
> Thank you for your support, sorry for the disturbing.
>
> Cordialement/ Best regards
> Patrice B
>
>
> -----Message d'origine-----
> De : Daniel-Constantin Mierla [mailto:miconda at gmail.com]
> Envoy? : vendredi 31 ao?t 2012 17:11
> ? : miconda at gmail.com; SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
> Cc : BODEVEN Patrice RD-CORE
> Objet : Re: [SR-Users] Kamailio - Nonce validity between Register and Invite
>
> Hello,
>
>
> On 8/31/12 5:08 PM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> I will look over the code, I just wanted to point that it is safer to
>> print by size in this case, otherwise it can get coredump, even if the
>> printed characters are not ascii, like:
>>
>> DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n",
>> &b_nonce.n.md5_1[0], 16);
> actually the size has to be first, like:
>
> DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %.*s)\n", 16,
> &b_nonce.n.md5_1[0]);
>
> Cheers,
> Daniel
>
>>
>> Otherwise DBG(..) goes on until finds '\0', which can be in a memory
>> zone used by another process.
>>
>> Cheers,
>> Daniel
>>
>> On 8/31/12 4:09 PM, patrice.bodeven at orange.com wrote:
>>> Hello
>>>
>>> The return code is 2.
>>>
>>> The problem seems due to "nonce.c"
>>> if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) {
>>> xxx
>>> }
>>>
>>> So I put
>>>
>>>      DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce %s)\n",
>>> &b_nonce.n.md5_1[0]);
>>>      DBG("auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2 %s)\n",
>>> &b_nonce2.n.md5_1[0]);
>>>      return 2;
>>> }
>>>
>>> The result into log is :
>>>
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> registrar [lookup.c:229]: '+33296053686' found in usrloc
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> auth [nonce.c:473]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce
>>> ????\?X`#035~MT~\?Ul~X~Sb~@???#025n?| ???Zo#032#Y?P~@??#010)
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> auth [nonce.c:474]: auth:pre_auth:api.c:nonce.c Invalid nonce (nonce2
>>> r?7B??| {?9R~R?FS?~Sb~@???#025n?|
>>> ???Zo#032#Y?P~@??#010PA#027#030P@??????\?X`#035~MT~\?Ul~X~Sb~@???#025n?|
>>> ???Zo#032#Y?P~@??#010)
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>>> Aug 31 13:58:25 openims-kamailio-1 /usr/sbin/kamailio[7824]: DEBUG:
>>> auth [api.c:148]: auth:pre_auth: Invalid nonce value received (ret 2)
>>>
>>> Cordialement/ Best regards
>>> Patrice B
>>>
>>> -----Message d'origine-----
>>> De : Daniel-Constantin Mierla [mailto:miconda at gmail.com]
>>> Envoy? : vendredi 31 ao?t 2012 10:35
>>> ? : SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>>> Users Mailing List
>>> Cc : BODEVEN Patrice RD-CORE
>>> Objet : Re: [SR-Users] Kamailio - Nonce validity between Register and
>>> Invite
>>>
>>> Hello,
>>>
>>> On 8/30/12 10:43 AM, patrice.bodeven at orange.com wrote:
>>>> Hello,
>>>>
>>>> Sorry, but i am blocked on the nonce reuse between the Register and
>>>> the INVITE even with the info provided in debug mode.
>>>>     My understanding is Nonce should be valid between SIP methods
>>>> until the nonce is valid in the time.
>>>> But I don't know how the Nonce is stored and how Kamailio is looking
>>>> for the nonce into the memory !
>>>>
>>>> I did a test on Register => nonce reuse is ok.
>>>> I did a test on INVITE => nonce reuse of Register or on previous
>>>> INVITE doesn't work !
>>>>
>>>> But based on the site indicated below and the SIPP (generate the
>>>> INVITE with the Nonce of Register), the result is
>>>> auth_db [authorize.c:271]: realm value [openims-kamailio-1.mycluster]
>>>> auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1
>>>> auth [api.c:147]: auth:pre_auth: Invalid nonce value received
>>>>
>>>> But there is no explanation why the nonce is invalid.
>>> can you apply the following patch:
>>>
>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blobdiff;f=modules/auth/api.c;h=c79de5c9dd6cd3e176c05d836c7b0a4275f8a3d2;hp=2ee39a7a6cbcce741365dac97ae8a3db8c196b0c;hb=494b383edde7a2d193c220f3117506e4cc95932f;hpb=e4ecf49add0e62330e1db071106806e35e9b078a
>>>
>>>
>>> It practically adds the return code in the log message, that should give
>>> better leads to why the nonce is considered invalid, allowing to track
>>> back quicker in the source code.
>>>
>>> Cheers,
>>> Daniel
>>>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From jh at tutpro.com  Fri Aug 31 20:59:33 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 21:59:33 +0300
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <5040FFE4.6040608@gmail.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
	<6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
	<20544.62363.353350.416390@siika.tutpro.com>
	<5040FFE4.6040608@gmail.com>
Message-ID: <20545.2453.843833.599491@siika.tutpro.com>

Daniel-Constantin Mierla writes:

> Can you set $var(plan) to a static string value before dp_translate() 
> just to see if the value is updated in the function?

i now have:

    xlog("L_INFO", "Translating <$var(calling_number)> based on dpid 4\n");
    $var(plan) = "something";
    if (dp_translate("4", "$var(calling_number)")) {
        xlog("L_INFO", "Found attrs <$var(plan)>\n");

and get:

Aug 31 21:57:21 wheezy2 /usr/sbin/sip-proxy[9733]: INFO: Translating <+35892345670> based on dpid 4
Aug 31 21:57:21 wheezy2 /usr/sbin/sip-proxy[9733]: INFO: Found attrs <598400320>

so looks like something happens to the pvar, but not the right thing.

-- juha


From jh at tutpro.com  Fri Aug 31 21:08:40 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 22:08:40 +0300
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <20545.2453.843833.599491@siika.tutpro.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
	<6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
	<20544.62363.353350.416390@siika.tutpro.com>
	<5040FFE4.6040608@gmail.com>
	<20545.2453.843833.599491@siika.tutpro.com>
Message-ID: <20545.3000.421173.522314@siika.tutpro.com>


> Aug 31 21:57:21 wheezy2 /usr/sbin/sip-proxy[9733]: INFO: Translating <+35892345670> based on dpid 4
> Aug 31 21:57:21 wheezy2 /usr/sbin/sip-proxy[9733]: INFO: Found attrs <598400320>

the result in attrs is pretty much random.  sometimes i get the right
result and sometimes not.  it does not seem to matter if i set
$var(plan) to 0, "", or some other string value.  i have got correct and
incorrect results with all combinations.

-- juha



From jh at tutpro.com  Fri Aug 31 21:20:46 2012
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 31 Aug 2012 22:20:46 +0300
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <20545.3000.421173.522314@siika.tutpro.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
	<6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
	<20544.62363.353350.416390@siika.tutpro.com>
	<5040FFE4.6040608@gmail.com>
	<20545.2453.843833.599491@siika.tutpro.com>
	<20545.3000.421173.522314@siika.tutpro.com>
Message-ID: <20545.3726.486945.550463@siika.tutpro.com>

other weird thing is that when i tried to add more debug output to
dialplan module and compiled dialplan module separately on another host
(the same one where the debian package installed on this test host has
been built) and then copied it to the host (virtual machine) where these
tests are done, i cannot start the proxy with the separately compiled
dialplan.so.  i get to syslog:

0(11244) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib/sip-proxy/modules/dialplan.so>: libkmi.so.1: cannot open shared object file: No such file or directory

when the file definitely exists:

/usr/lib/sip-proxy/modules# ls -ls dialplan.so
148 -rwxr-xr-x 1 jh jh 151353 Aug 31 22:10 dialplan.so

may be all this hassle is somehow related to trying to run my sip proxy
on a Virtualbox virtual machine?

-- juha


From spencer at 5ninesolutions.com  Fri Aug 31 22:07:19 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 31 Aug 2012 14:07:19 -0600
Subject: [SR-Users] Rtpproxy re-INVITE handling
In-Reply-To: <50410396.2050805@gmail.com>
References: <2998a231-2f38-4c5f-88b9-e8f0e89229ff@blur>
	<503F1B68.3060107@gmail.com>
	<6BA29812-8FD9-49FC-A6C4-FE9B4FAA53EE@5ninesolutions.com>
	<503FB817.4060009@gmail.com>
	<B03A90C3-6658-4F80-94BE-7EBB7D39287E@5ninesolutions.com>
	<50407B8C.6090503@gmail.com>
	<76e2de03d32bc5ec4e07e04bfebb4bdf@mail.5ninesolutions.com>
	<50410396.2050805@gmail.com>
Message-ID: <c02644a5-9d26-49fb-9fc6-44d40156963f@blur>

Completely understandable :-).  I'm currently our of the office until Wednesday but I will collect logs with a trace upon my return.
 

Connected by DROID on Verizon Wireless

-----Original message-----
From: Daniel-Constantin Mierla <miconda at gmail.com>
To: Spencer Thomason <spencer at 5ninesolutions.com>
Cc: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users at lists.sip-router.org>
Sent: Fri, Aug 31, 2012 18:34:11 GMT+00:00
Subject: Re: [SR-Users] Rtpproxy re-INVITE handling

Hello,

the command to rtpproxy for the reply seem to miss the to-tag, can you 
grab the ngrep trace for such call and the logs for processing it? 
Having the logs from a different call than the ngrep trace you posted on 
pastebin is not helping much.

Cheers,
Daniel


On 8/31/12 6:49 PM, Spencer Thomason wrote:
>
> Yes,
>
> The request (re-INVITE):
>
> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received 
> command "25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 184.170.249.3 32122 
> 3ae1Dvgr5vmeg;1 199857477;1"
> Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: adding 
> strong flag to existing session, new=1/0/0
> Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup on 
> ports 55324/46010, session timer restarted
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply 
> "25778_11 46010 184.170.249.8#012"
>
> The reply:
>
> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received 
> command "25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 71.104.248.48 6016 
> 3ae1Dvgr5vmeg;1"
> Aug 30 22:38:59 sip rtpproxy[25671]: INFO:handle_command: lookup 
> request failed: session 1952045641-6076-15 at BA.FJ.B.CFC, tags 
> 3ae1Dvgr5vmeg;1/NONE not found
> Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: sending reply 
> "25778_12 0 184.170.249.8#012"
> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: rtpproxy 
> [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy
>
> I was able to get it working correctly by reworking the config like 
> the 3.1 branch by using rtpproxy_offer instead of force_rtp_proxy. 
>  When I attempted to use rtpproxy_answer in the reply route, I was 
> getting the same lookup request failed error from rtpproxy.  In the 
> request and reply, the tags change.  Could this be the reason that the 
> session lookup is failing?  If I use rtpproxy_offer in both the 
> request and reply, everything works correctly.  Is there any 
> consequence to doing this?
>
> Thanks,
>
> Spencer
>
> On 31.08.2012 01:53, Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> On 8/31/12 3:41 AM, Spencer Thomason wrote:
>>> Hi Daniel, I can confirm that rtpproxy_manage is called. See: 
>>> http://pastebin.com/ZVXjK9ry I'm seeing ERROR: rtpproxy 
>>> [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy in the 
>>> logs when processing the 200OK in the re-INVITE. I've included a 
>>> debug level log from rtpproxy in the log as well.
>> this can happen because the rtpproxy was not engaged for the request,
>> but only for the reply.
>>
>> As you say, the logs are for the 200OK, what about the ones for request,
>> is rtpproxy called there?
>>
>> Cheers,
>> Daniel
>>> When handling the re-INVITE there is: ? Aug 30 22:38:59 sip 
>>> /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
>>> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage ? Aug 30 
>>> 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command 
>>> "25778_11 U 1952045641-6076-15 at BA.FJ.B.CFC 
>>> <mailto:1952045641-6076-15 at BA.FJ.B.CFC> 184.170.249.3 32122 
>>> 3ae1Dvgr5vmeg;1 199857477;1" ? Aug 30 22:38:59 sip rtpproxy[25671]: 
>>> INFO:handle_command: adding strong flag to existing session, 
>>> new=1/0/0 ? Aug 30 22:38:59 sip rtpproxy[25671]: 
>>> INFO:handle_command: lookup on ports 55324/46010, session timer 
>>> restarted ? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: 
>>> sending reply "25778_11 46010 184.170.249.8#012" but the 200OK: ? 
>>> Aug 30 22:38:59 sip /usr/sbin/kamailio[25778]: ERROR: *** cfgtrace: 
>>> c=[/etc/kamailio/kamailio.cfg] l=471 a=25 n=rtpproxy_manage ? Aug 30 
>>> 22:38:59 sip rtpproxy[25671]: DBUG:handle_command: received command 
>>> "25778_12 L 1952045641-6076-15 at BA.FJ.B.CFC 
>>> <mailto:1952045641-6076-15 at BA.FJ.B.CFC> 71.104.248.48 6016 
>>> 3ae1Dvgr5vmeg;1" ? Aug 30 22:38:59 sip rtpproxy[25671]: 
>>> INFO:handle_command: lookup request failed: session 
>>> 1952045641-6076-15 at BA.FJ.B.CFC 
>>> <mailto:1952045641-6076-15 at BA.FJ.B.CFC>, tags 3ae1Dvgr5vmeg;1/NONE 
>>> not found ? Aug 30 22:38:59 sip rtpproxy[25671]: DBUG:doreply: 
>>> sending reply "25778_12 0 184.170.249.8#012" ? Aug 30 22:38:59 sip 
>>> /usr/sbin/kamailio[25778]: ERROR: rtpproxy [rtpproxy.c:2260]: 
>>> incorrect port 0 in reply from rtp proxy I'm not familiar with the 
>>> rtpproxy commands to know why it cannot locate the session. Thanks 
>>> for your assistance, Spencer On Aug 30, 2012, at 11:59 AM, 
>>> Daniel-Constantin Mierla wrote:
>>>> Hello, I could not spot by quick eye checking what could happen 
>>>> there, the best is to use the debugger module with cfg_trace 
>>>> parameter set and check the execution trace to see what actions of 
>>>> the configuration file are executed and be sure the rtpproxy is 
>>>> called or not. You can post the execution trace here if you need 
>>>> further help with it. Cheers, Daniel On 8/30/12 7:40 PM, Spencer 
>>>> Thomason wrote:
>>>>> Hi Daniel, Thanks for your help with this. Here is a trace: 
>>>>> http://pastebin.com/pXeFbwBz I see the nat=yes parameter added to 
>>>>> the Route header. I've posted the script here: 
>>>>> http://pastebin.com/2qwHYHvjForgive my ignorance, I can't tell 
>>>>> what I'm doing wrong. Thanks! Spencer On Aug 30, 2012, at 12:51 
>>>>> AM, Daniel-Constantin Mierla wrote:
>>>>>> Hello, if your config it is based on the default one, the Route 
>>>>>> header for within dialog requests is marked by a parameter, 
>>>>>> nat=yes, that is used to decide whether to do rtpproxy or not. 
>>>>>> So, if you have a custom config, check the default one for the 
>>>>>> nat traversal handling part. Cheers, Daniel On 8/30/12 12:39 AM, 
>>>>>> Spencer Thomason wrote:
>>>>>>> Hello, I'm using Kamailio 3.2.4 for NAT traversal using 
>>>>>>> rtpproxy_manage() in a largely stock script. Everything works 
>>>>>>> great until the far end (on a public ip) sends a t.38 re-INVITE. 
>>>>>>> The 200OK from the NATed UAC then doesn't trigger rtpproxy and 
>>>>>>> the private IP in the sdp causes the fax to fail. Any help 
>>>>>>> handling these re-INVITEs would be greatly appreciated. I'm 
>>>>>>> happy to post traces if that helps describe the situation. The 
>>>>>>> network topology looks like this: NATed UAC -> Kamailio on a 
>>>>>>> public IP -> UAS on a public IP Thanks in advance, Spencer 
>>>>>>> Connected by DROID on Verizon Wireless 
>>>>>>> _______________________________________________ SIP Express 
>>>>>>> Router (SER) and Kamailio (OpenSER) - sr-users mailing list 
>>>>>>> sr-users at lists.sip-router.org 
>>>>>>> <mailto:sr-users at lists.sip-router.org> 
>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>> -- Daniel-Constantin Mierla - http://www.asipto.com 
>>>>>> http://twitter.com/#!/miconda - 
>>>>>> http://www.linkedin.com/in/miconda Kamailio Advanced Training, 
>>>>>> Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
>>>> -- Daniel-Constantin Mierla - http://www.asipto.com 
>>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda 
>>>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - 
>>>> http://asipto.com/u/kat
>>> _______________________________________________ SIP Express Router 
>>> (SER) and Kamailio (OpenSER) - sr-users mailing list 
>>> sr-users at lists.sip-router.org <mailto:sr-users at lists.sip-router.org> 
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From miconda at gmail.com  Fri Aug 31 22:37:17 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 31 Aug 2012 22:37:17 +0200
Subject: [SR-Users] weird dp_translate result
In-Reply-To: <20545.3726.486945.550463@siika.tutpro.com>
References: <20544.58129.485514.359848@siika.tutpro.com>
	<6CEB1BB8-62EB-454B-ADB1-24E9219B9CB4@gmail.com>
	<20544.62363.353350.416390@siika.tutpro.com>
	<5040FFE4.6040608@gmail.com>
	<20545.2453.843833.599491@siika.tutpro.com>
	<20545.3000.421173.522314@siika.tutpro.com>
	<20545.3726.486945.550463@siika.tutpro.com>
Message-ID: <5041207D.9070903@gmail.com>


On 8/31/12 9:20 PM, Juha Heinanen wrote:
> other weird thing is that when i tried to add more debug output to
> dialplan module and compiled dialplan module separately on another host
> (the same one where the debian package installed on this test host has
> been built) and then copied it to the host (virtual machine) where these
> tests are done, i cannot start the proxy with the separately compiled
> dialplan.so.  i get to syslog:
>
> 0(11244) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/lib/sip-proxy/modules/dialplan.so>: libkmi.so.1: cannot open shared object file: No such file or directory
>
> when the file definitely exists:
>
> /usr/lib/sip-proxy/modules# ls -ls dialplan.so
> 148 -rwxr-xr-x 1 jh jh 151353 Aug 31 22:10 dialplan.so
The complain seems to be regarding missing of libkmi, can you check that 
is installed as well?

>
> may be all this hassle is somehow related to trying to run my sip proxy
> on a Virtualbox virtual machine?

Do you have any possibility to test it where it compiles? You can even 
run from source tree, without a need to install, connecting to the 
database server where dialplan table is located -- just to avoid hunting 
ghost issues in the code.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat