From ozren.lapcevic at gmail.com  Wed Jun  1 10:48:38 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Wed, 1 Jun 2011 10:48:38 +0200
Subject: [SR-Users] IMC module unresolved issues
Message-ID: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>

Hi,

did anyone run into similar problems or have successfully set up IMC module
with mysql? Can you please share your know-how on the subject?

Besides issue described below, I also had several other issues, but I think
they all might be related to the fact that my rooms and members are only
stored in cache, but not in the database. To repeat, this is really strange
behavior because all other modules that I'm using write to the database
without a problem.

Best regards
Ozren


On Mon, May 30, 2011 at 6:28 PM, Ozren Lapcevic <ozren.lapcevic at gmail.com>wrote:

> Hi all,
>
> I'm trying to configure IMC module and get it to work properly but I've run
> into problems with saving IMC module data to mysql database.
>
> I'm using following code provided in IMC module documentation:
>
> modparam("imc", "db_url", DBURL)
>
> ## route block ###
>         if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~
> "sip:chat-manager@")) {
>                 xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n");
>
>                 if(imc_manager())
>                         sl_send_reply("200", "ok - imc_manager()");
>                 else
>                         sl_send_reply("500", "command error");
>         exit;
>         }
>
> I am creating chat room with following message sent to chat-manager at abc.hr:
> "#create chat-111"
>
> DEBUG log returns following:
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-manager at abc.hr]
> B [#create chat-111]
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc.c:525]: found command
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_cmd.c:146]: command: [create]
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111]
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_cmd.c:171]: new room [chat-111]
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111 at abc.hr
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz at abc.hr
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_mng.c:326]: [user]= oz
> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
> DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [
> sip:oz at abc.hr]
>
> "./kamctl fifo imc_list_rooms" returns following:
> ROOM::  URI=sip:chat-111 at abc.hr MEMBERS=0 OWNER=sip:oz at abc.hr
>
> "./kamctl fifo imc_list_members sip:chat-111 at abc.hr" returns following:
> ROOM:: sip:chat-111 at abc.hr NR_OF_MEMBERS=1
>         MEMBER:: sip:oz at abc.hr
>
> However, that data is not persisted/saved in the imc_rooms and imc_members
> tables in the database. Any clues what might be the problem here?
>
> Any help is greatly appreciated.
>
> Best regards,
> Ozren Lapcevic
>
> PS
> I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql,
> multidomain, usrlocdb, auth and NAT. I can register clients, establish
> calls, send IMs, etc. imc_rooms and imc_members tables were created when
> kamailio was installed. I've also checked imc-create.sql script and tables
> are correctly created.
>
> The same DBURL used for imc is also used for auth_db, domain and usrloc
> modules and I haven't experienced any problems with those modules and
> database connection/persistence. domain, location, version and subscriber
> tables are correctly populated.
>
> I'm not getting any errors from imc_manager(), it returns 200 OK. I can't
> trace any errors in DEBUG related to problems with mysql database.
>
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From miconda at gmail.com  Wed Jun  1 12:41:01 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Jun 2011 12:41:01 +0200
Subject: [SR-Users] IMC module unresolved issues
In-Reply-To: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>
References: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>
Message-ID: <4DE6173D.6000409@gmail.com>

Hello,

the module is writing to database only at shutdown, otherwise it keeps 
everything in cache -- just in case you expected to see some records in 
database at runtime. However, I think there was an issue that even at 
shutdown the module didn't write back to database. Please fetch the 
latest git branch 3.1 and try again for this purpose.

Thanks,
Daniel

On 6/1/11 10:48 AM, Ozren Lapcevic wrote:
> Hi,
>
> did anyone run into similar problems or have successfully set up IMC 
> module with mysql? Can you please share your know-how on the subject?
>
> Besides issue described below, I also had several other issues, but I 
> think they all might be related to the fact that my rooms and members 
> are only stored in cache, but not in the database. To repeat, this is 
> really strange behavior because all other modules that I'm using write 
> to the database without a problem.
>
> Best regards
> Ozren
>
>
> On Mon, May 30, 2011 at 6:28 PM, Ozren Lapcevic 
> <ozren.lapcevic at gmail.com <mailto:ozren.lapcevic at gmail.com>> wrote:
>
>     Hi all,
>
>     I'm trying to configure IMC module and get it to work properly but
>     I've run into problems with saving IMC module data to mysql database.
>
>     I'm using following code provided in IMC module documentation:
>
>     modparam("imc", "db_url", DBURL)
>
>     ## route block ###
>             if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") ||
>     (uri=~ "sip:chat-manager@")) {
>                     xlog("L_INFO", "going to imc: F [$fu] T [$tu] B
>     [$rb]\n");
>
>                     if(imc_manager())
>                             sl_send_reply("200", "ok - imc_manager()");
>                     else
>                             sl_send_reply("500", "command error");
>             exit;
>             }
>
>     I am creating chat room with following message sent to
>     chat-manager at abc.hr <mailto:chat-manager at abc.hr>: "#create chat-111"
>
>     DEBUG log returns following:
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: INFO: <script>:
>     going to imc: F [sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>] T
>     [sip:chat-manager at abc.hr <mailto:sip%3Achat-manager at abc.hr>] B
>     [#create chat-111]
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc.c:525]: found command
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_cmd.c:146]: command: [create]
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_cmd.c:151]: parameter 0=[chat-111]
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_cmd.c:171]: new room [chat-111]
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_cmd.c:186]: added room uri= sip:chat-111 at abc.hr
>     <mailto:sip%3Achat-111 at abc.hr>
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_mng.c:322]: [uri]= sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_mng.c:326]: [user]= oz
>     May 30 18:01:23 ser-test
>     /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>     [imc_cmd.c:196]: added the owner as the first member
>     [sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>]
>
>     "./kamctl fifo imc_list_rooms" returns following:
>     ROOM::  URI=sip:chat-111 at abc.hr <mailto:sip%3Achat-111 at abc.hr>
>     MEMBERS=0 OWNER=sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>
>
>     "./kamctl fifo imc_list_members sip:chat-111 at abc.hr
>     <mailto:sip%3Achat-111 at abc.hr>" returns following:
>     ROOM:: sip:chat-111 at abc.hr <mailto:sip%3Achat-111 at abc.hr>
>     NR_OF_MEMBERS=1
>             MEMBER:: sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>
>
>     However, that data is not persisted/saved in the imc_rooms and
>     imc_members tables in the database. Any clues what might be the
>     problem here?
>
>     Any help is greatly appreciated.
>
>     Best regards,
>     Ozren Lapcevic
>
>     PS
>     I'm runing Kamailio 3.1.4 with default cfg file with enabled
>     mysql, multidomain, usrlocdb, auth and NAT. I can register
>     clients, establish calls, send IMs, etc. imc_rooms and imc_members
>     tables were created when kamailio was installed. I've also checked
>     imc-create.sql script and tables are correctly created.
>
>     The same DBURL used for imc is also used for auth_db, domain and
>     usrloc modules and I haven't experienced any problems with those
>     modules and database connection/persistence. domain, location,
>     version and subscriber tables are correctly populated.
>
>     I'm not getting any errors from imc_manager(), it returns 200 OK.
>     I can't trace any errors in DEBUG related to problems with mysql
>     database.
>
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From klaus.mailinglists at pernau.at  Wed Jun  1 14:33:48 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 01 Jun 2011 14:33:48 +0200
Subject: [SR-Users] Kamailio Tables with index
Message-ID: <4DE631AC.5040905@pernau.at>

Hi!

With Openser 1.2 all the tables got an auto-increment-index column. Is
this index column just for layout/performance purposes or is it also
used somewhere insides the Kamailio code?

If not (what I expect), then Kamailio 3.1 should still work with tables
without those columns - correct?

Thanks
Klaus


From ozren.lapcevic at gmail.com  Wed Jun  1 15:33:36 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Wed, 1 Jun 2011 15:33:36 +0200
Subject: [SR-Users] IMC module unresolved issues
In-Reply-To: <4DE6173D.6000409@gmail.com>
References: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>
	<4DE6173D.6000409@gmail.com>
Message-ID: <BANLkTikmtRezEkXE+wAz1oYxSV=FXGR65w@mail.gmail.com>

Hi Daniel,

thanks for quick response and clarification, it helped a lot.

I was indeed expecting records in database at runtime, but until your latest
updates to imc module, the module didn't write to database even at shutdown.
Now, I can see records in the database when I shutdown Kamailio. When I
start up Kamailio it populates cache based on records from database and
deletes records in the database.

Best regards,
Ozren


On Wed, Jun 1, 2011 at 12:41 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> the module is writing to database only at shutdown, otherwise it keeps
> everything in cache -- just in case you expected to see some records in
> database at runtime. However, I think there was an issue that even at
> shutdown the module didn't write back to database. Please fetch the latest
> git branch 3.1 and try again for this purpose.
>
> Thanks,
> Daniel
>
> On 6/1/11 10:48 AM, Ozren Lapcevic wrote:
>
> Hi,
>
> did anyone run into similar problems or have successfully set up IMC module
> with mysql? Can you please share your know-how on the subject?
>
> Besides issue described below, I also had several other issues, but I think
> they all might be related to the fact that my rooms and members are only
> stored in cache, but not in the database. To repeat, this is really strange
> behavior because all other modules that I'm using write to the database
> without a problem.
>
> Best regards
> Ozren
>
>
> On Mon, May 30, 2011 at 6:28 PM, Ozren Lapcevic <ozren.lapcevic at gmail.com>wrote:
>
>> Hi all,
>>
>> I'm trying to configure IMC module and get it to work properly but I've
>> run into problems with saving IMC module data to mysql database.
>>
>> I'm using following code provided in IMC module documentation:
>>
>> modparam("imc", "db_url", DBURL)
>>
>> ## route block ###
>>         if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~
>> "sip:chat-manager@")) {
>>                 xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n");
>>
>>                 if(imc_manager())
>>                         sl_send_reply("200", "ok - imc_manager()");
>>                 else
>>                         sl_send_reply("500", "command error");
>>         exit;
>>         }
>>
>> I am creating chat room with following message sent to
>> chat-manager at abc.hr: "#create chat-111"
>>
>> DEBUG log returns following:
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [
>> sip:chat-manager at abc.hr] B [#create chat-111]
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc.c:525]: found command
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_cmd.c:146]: command: [create]
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111]
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_cmd.c:171]: new room [chat-111]
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111 at abc.hr
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz at abc.hr
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_mng.c:326]: [user]= oz
>> May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]:
>> DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [
>> sip:oz at abc.hr]
>>
>> "./kamctl fifo imc_list_rooms" returns following:
>> ROOM::  URI=sip:chat-111 at abc.hr MEMBERS=0 OWNER=sip:oz at abc.hr
>>
>> "./kamctl fifo imc_list_members sip:chat-111 at abc.hr" returns following:
>> ROOM:: sip:chat-111 at abc.hr NR_OF_MEMBERS=1
>>         MEMBER:: sip:oz at abc.hr
>>
>> However, that data is not persisted/saved in the imc_rooms and imc_members
>> tables in the database. Any clues what might be the problem here?
>>
>> Any help is greatly appreciated.
>>
>> Best regards,
>> Ozren Lapcevic
>>
>> PS
>> I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql,
>> multidomain, usrlocdb, auth and NAT. I can register clients, establish
>> calls, send IMs, etc. imc_rooms and imc_members tables were created when
>> kamailio was installed. I've also checked imc-create.sql script and tables
>> are correctly created.
>>
>> The same DBURL used for imc is also used for auth_db, domain and usrloc
>> modules and I haven't experienced any problems with those modules and
>> database connection/persistence. domain, location, version and subscriber
>> tables are correctly populated.
>>
>> I'm not getting any errors from imc_manager(), it returns 200 OK. I can't
>> trace any errors in DEBUG related to problems with mysql database.
>>
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From miconda at gmail.com  Wed Jun  1 16:18:52 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Jun 2011 16:18:52 +0200
Subject: [SR-Users] IMC module unresolved issues
In-Reply-To: <BANLkTikmtRezEkXE+wAz1oYxSV=FXGR65w@mail.gmail.com>
References: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>
	<4DE6173D.6000409@gmail.com>
	<BANLkTikmtRezEkXE+wAz1oYxSV=FXGR65w@mail.gmail.com>
Message-ID: <4DE64A4C.7090201@gmail.com>

Hello,

On 6/1/11 3:33 PM, Ozren Lapcevic wrote:
> Hi Daniel,
>
> thanks for quick response and clarification, it helped a lot.
>
> I was indeed expecting records in database at runtime, but until your 
> latest updates to imc module, the module didn't write to database even 
> at shutdown. Now, I can see records in the database when I shutdown 
> Kamailio. When I start up Kamailio it populates cache based on records 
> from database and deletes records in the database.
great, thanks for reporting back it's working.

Any issues still going on with the module?

Best regards,
Daniel

>
> Best regards,
> Ozren
>
>
> On Wed, Jun 1, 2011 at 12:41 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     the module is writing to database only at shutdown, otherwise it
>     keeps everything in cache -- just in case you expected to see some
>     records in database at runtime. However, I think there was an
>     issue that even at shutdown the module didn't write back to
>     database. Please fetch the latest git branch 3.1 and try again for
>     this purpose.
>
>     Thanks,
>     Daniel
>
>     On 6/1/11 10:48 AM, Ozren Lapcevic wrote:
>>     Hi,
>>
>>     did anyone run into similar problems or have successfully set up
>>     IMC module with mysql? Can you please share your know-how on the
>>     subject?
>>
>>     Besides issue described below, I also had several other issues,
>>     but I think they all might be related to the fact that my rooms
>>     and members are only stored in cache, but not in the database. To
>>     repeat, this is really strange behavior because all other modules
>>     that I'm using write to the database without a problem.
>>
>>     Best regards
>>     Ozren
>>
>>
>>     On Mon, May 30, 2011 at 6:28 PM, Ozren Lapcevic
>>     <ozren.lapcevic at gmail.com <mailto:ozren.lapcevic at gmail.com>> wrote:
>>
>>         Hi all,
>>
>>         I'm trying to configure IMC module and get it to work
>>         properly but I've run into problems with saving IMC module
>>         data to mysql database.
>>
>>         I'm using following code provided in IMC module documentation:
>>
>>         modparam("imc", "db_url", DBURL)
>>
>>         ## route block ###
>>                 if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@")
>>         || (uri=~ "sip:chat-manager@")) {
>>                         xlog("L_INFO", "going to imc: F [$fu] T [$tu]
>>         B [$rb]\n");
>>
>>                         if(imc_manager())
>>                                 sl_send_reply("200", "ok -
>>         imc_manager()");
>>                         else
>>                                 sl_send_reply("500", "command error");
>>                 exit;
>>                 }
>>
>>         I am creating chat room with following message sent to
>>         chat-manager at abc.hr <mailto:chat-manager at abc.hr>: "#create
>>         chat-111"
>>
>>         DEBUG log returns following:
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: INFO:
>>         <script>: going to imc: F [sip:oz at abc.hr
>>         <mailto:sip%3Aoz at abc.hr>] T [sip:chat-manager at abc.hr
>>         <mailto:sip%3Achat-manager at abc.hr>] B [#create chat-111]
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc.c:525]: found command
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_cmd.c:146]: command: [create]
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_cmd.c:151]: parameter 0=[chat-111]
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_cmd.c:171]: new room [chat-111]
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_cmd.c:186]: added room uri= sip:chat-111 at abc.hr
>>         <mailto:sip%3Achat-111 at abc.hr>
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_mng.c:322]: [uri]= sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_mng.c:326]: [user]= oz
>>         May 30 18:01:23 ser-test
>>         /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc
>>         [imc_cmd.c:196]: added the owner as the first member
>>         [sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>]
>>
>>         "./kamctl fifo imc_list_rooms" returns following:
>>         ROOM::  URI=sip:chat-111 at abc.hr
>>         <mailto:sip%3Achat-111 at abc.hr> MEMBERS=0 OWNER=sip:oz at abc.hr
>>         <mailto:sip%3Aoz at abc.hr>
>>
>>         "./kamctl fifo imc_list_members sip:chat-111 at abc.hr
>>         <mailto:sip%3Achat-111 at abc.hr>" returns following:
>>         ROOM:: sip:chat-111 at abc.hr <mailto:sip%3Achat-111 at abc.hr>
>>         NR_OF_MEMBERS=1
>>                 MEMBER:: sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>
>>
>>         However, that data is not persisted/saved in the imc_rooms
>>         and imc_members tables in the database. Any clues what might
>>         be the problem here?
>>
>>         Any help is greatly appreciated.
>>
>>         Best regards,
>>         Ozren Lapcevic
>>
>>         PS
>>         I'm runing Kamailio 3.1.4 with default cfg file with enabled
>>         mysql, multidomain, usrlocdb, auth and NAT. I can register
>>         clients, establish calls, send IMs, etc. imc_rooms and
>>         imc_members tables were created when kamailio was installed.
>>         I've also checked imc-create.sql script and tables are
>>         correctly created.
>>
>>         The same DBURL used for imc is also used for auth_db, domain
>>         and usrloc modules and I haven't experienced any problems
>>         with those modules and database connection/persistence.
>>         domain, location, version and subscriber tables are correctly
>>         populated.
>>
>>         I'm not getting any errors from imc_manager(), it returns 200
>>         OK. I can't trace any errors in DEBUG related to problems
>>         with mysql database.
>>
>>
>>
>>
>>     _______________________________________________
>>     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://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

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From ozren.lapcevic at gmail.com  Wed Jun  1 16:25:49 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Wed, 1 Jun 2011 16:25:49 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
Message-ID: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>

Hi,

there is a bug in IMC module that crashes Kamailio.

Here is the scenario: One user (pero at abc.hr) creates private chat room (
chat-999 at abc.hr). Other user (oz at abc.hr) tries to join the
chat-999 at abc.hrby sending #join. Kamailio crashes at that point. Here
is the log:

Best regards,
Ozren


Jun  1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]: segfault
at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in imc.so[7f28f89cf000+17000]
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-999 at abc.hr] B
[#join]
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: <core> [receive.c:186]: preparing to run routing scripts...
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
DEBUG: imc [imc.c:525]: found command
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: maxfwd [mf_funcs.c:85]: value = 70
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
DEBUG: imc [imc_cmd.c:146]: command: [join]
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body :
content_length=223
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
DEBUG: imc [imc_cmd.c:292]: found room [chat-999]
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: <core> [parser/msg_parser.c:103]: found end of header
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user
[oz]
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
DEBUG: <core> [parser/msg_parser.c:630]: SIP Request:
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
ALERT: <core> [main.c:744]: core was not generated
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82,
161.53.255.82, 0)
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
DEBUG: <core> [parser/msg_parser.c:632]:  method:  <MESSAGE>
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]:
INFO: <core> [main.c:807]: INFO: signal 15 received
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
DEBUG: imc [imc.c:709]: room 3 chat-777
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
DEBUG: imc [imc.c:709]: room 11 chat-888
Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
DEBUG: imc [imc.c:709]: room 13 chat-999
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From miconda at gmail.com  Wed Jun  1 16:29:18 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 01 Jun 2011 16:29:18 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
In-Reply-To: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
References: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
Message-ID: <4DE64CBE.3030607@gmail.com>

Hello,

can you get the core dump and send the backtrace?

Be sure you have 'ulimit -c unlimited', working directory (-w option) 
set to writable path and you don't run it in chowner (no -u or -g 
parameters).

Cheers,
Daniel

On 6/1/11 4:25 PM, Ozren Lapcevic wrote:
> Hi,
>
> there is a bug in IMC module that crashes Kamailio.
>
> Here is the scenario: One user (pero at abc.hr <mailto:pero at abc.hr>) 
> creates private chat room (chat-999 at abc.hr <mailto:chat-999 at abc.hr>). 
> Other user (oz at abc.hr <mailto:oz at abc.hr>) tries to join the 
> chat-999 at abc.hr <mailto:chat-999 at abc.hr> by sending #join. Kamailio 
> crashes at that point. Here is the log:
>
> Best regards,
> Ozren
>
>
> Jun  1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]: 
> segfault at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in 
> imc.so[7f28f89cf000+17000]
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20118]: INFO: <script>: going 
> to imc: F [sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>] T 
> [sip:chat-999 at abc.hr <mailto:sip%3Achat-999 at abc.hr>] B [#join]
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> 
> [receive.c:186]: preparing to run routing scripts...
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc 
> [imc.c:525]: found command
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: maxfwd 
> [mf_funcs.c:85]: value = 70
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc 
> [imc_cmd.c:146]: command: [join]
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> 
> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=223
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc 
> [imc_cmd.c:292]: found room [chat-999]
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> 
> [parser/msg_parser.c:103]: found end of header
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20118]: ERROR: imc 
> [imc_cmd.c:317]: attept to join private room [chat-999] from user [oz]
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> 
> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> 
> [main.c:741]: child process 20118 exited by a signal 11
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> 
> [parser/parse_to.c:803]: end of header reached, state=29
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> 
> [parser/msg_parser.c:630]: SIP Request:
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> 
> [main.c:744]: core was not generated
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> 
> [msg_translator.c:204]: check_via_address(161.53.255.82, 161.53.255.82, 0)
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> 
> [parser/msg_parser.c:632]:  method: <MESSAGE>
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20113]: INFO: <core> 
> [main.c:756]: INFO: terminating due to SIGCHLD
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20114]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20115]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20116]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20117]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20122]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20124]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20125]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20129]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20128]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20127]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20123]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20126]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20120]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20130]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20131]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc 
> [imc.c:709]: room 3 chat-777
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc 
> [imc.c:709]: room 11 chat-888
> Jun  1 15:10:37 ser-test 
> /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc 
> [imc.c:709]: room 13 chat-999
>
>
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From henning.westerholt at 1und1.de  Wed Jun  1 16:37:41 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Wed, 1 Jun 2011 16:37:41 +0200
Subject: [SR-Users] Kamailio Tables with index
In-Reply-To: <4DE631AC.5040905@pernau.at>
References: <4DE631AC.5040905@pernau.at>
Message-ID: <201106011637.41166.henning.westerholt@1und1.de>

On Wednesday 01 June 2011, Klaus Darilion wrote:
> With Openser 1.2 all the tables got an auto-increment-index column. Is
> this index column just for layout/performance purposes or is it also
> used somewhere insides the Kamailio code?
> 
> If not (what I expect), then Kamailio 3.1 should still work with tables
> without those columns - correct?

Hi Klaus,

i think your assumption is correct, as they are some DB modules that don't 
have support for this auto_increment type.

Best regards,

Henning


From bruno.bresciani at gmail.com  Wed Jun  1 16:48:45 2011
From: bruno.bresciani at gmail.com (Bruno Bresciani)
Date: Wed, 1 Jun 2011 11:48:45 -0300
Subject: [SR-Users] has_tran_tmcbs function
Message-ID: <BANLkTi=Soj41PSG9q02+7ww_pF3MGhnV0g@mail.gmail.com>

Hi,

I can't understand the has_tran_tmcbs function... Can someone tell me how it
works?


Regards,
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From timo.reimann at 1und1.de  Wed Jun  1 17:12:37 2011
From: timo.reimann at 1und1.de (Timo Reimann)
Date: Wed, 1 Jun 2011 17:12:37 +0200
Subject: [SR-Users] has_tran_tmcbs function
In-Reply-To: <BANLkTi=Soj41PSG9q02+7ww_pF3MGhnV0g@mail.gmail.com>
References: <BANLkTi=Soj41PSG9q02+7ww_pF3MGhnV0g@mail.gmail.com>
Message-ID: <4DE656E5.8070100@1und1.de>

Hey Bruno,


On 01.06.2011 16:48, Bruno Bresciani wrote:
> I can't understand the has_tran_tmcbs function... Can someone tell me
> how it works?

The function checks if, for a given transaction T and a bitfield of
callbacks, any of the callbacks are registered for T.

It does so my logically ANDing the transaction bitfield (which
represents the set of registered callbacks) with the bitfield of
callbacks you would like to be checked. For each callback actually
registered, the corresponding pair of bits will yield one w.r.t. to the
AND operation. Therefore, if at least one of the given callbacks is
registered, the entire AND operation will result in a value greater
zero, thereby indicating that the transaction does have at least one of
the given callbacks registered.

For instance, if the bitfield of registered callbacks for T is 1011, and
it is to be checked whether the callback bitfield 0110 matches, the
outcome of 1011 & 0110 = 0010 > 0, meaning that at least one of the
given callbacks is registered (which, in this example, happens to be the
second transaction callback defined in t_hooks.h).


HTH,

--Timo


From marketing at cluecon.com  Wed Jun  1 00:18:47 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Tue, 31 May 2011 22:18:47 +0000
Subject: [SR-Users] ClueCon 2011 News and Notes
Message-ID: <0000013048233f42-c4fc8c94-e5a3-4e82-bf5b-745457df018d-000000@email.amazonses.com>

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From skchopperguy at gmail.com  Wed Jun  1 21:38:53 2011
From: skchopperguy at gmail.com (Skyler)
Date: Wed, 1 Jun 2011 12:38:53 -0700
Subject: [SR-Users] serweb & xSER 3.1
Message-ID: <040a01cc2093$8ae88f80$a0b9ae80$@com>

Hi all,

 

 Wondering if anyone has serweb working/functioning with Kamailio or SER 3.1
branch? If so which version of serweb are you using 2.0.0 or 2.1.0-dev1? Any
issues or tips for first time install/setup of serweb?

 

S.

 

 

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From lucasaa at gmail.com  Thu Jun  2 01:08:00 2011
From: lucasaa at gmail.com (Lucas Alvarez)
Date: Wed, 1 Jun 2011 20:08:00 -0300
Subject: [SR-Users] RTP traffic
In-Reply-To: <4DE52937.9090308@evaristesys.com>
References: <BANLkTin8O8WKnEXvEf2xg5kN3jXzNh5CLA@mail.gmail.com>
	<4DE52937.9090308@evaristesys.com>
Message-ID: <BANLkTimCgt8rPfv2LH5=aBz5gsETsgW5ZQ@mail.gmail.com>

Alex, thank you for the answer, I have installed rtpproxy and is running:

rtpproxy -l 66.152.18.221 -s udp:localhost:7722

this is my NAT kamailio.cfg:

#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org")

# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif


And in the route(NAT) I have changed the nat test:

*           if (!nat_uac_test("32"))*

This is because I want to do NAT only with those clients which source
signalling is not RFC1918 compliant.
Since I'm using a transparent proxy on the public ip. The problem I'm still
having is that the RTP traffic are routing to the private ip address of the
sip phone. I have been with this since a couple of days and I would
appreciate a little help.
Thanks ons advance,


Lucas
On Tue, May 31, 2011 at 2:45 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> On 05/31/2011 01:40 PM, Lucas Alvarez wrote:
>
>  Hi, I'm having a problem with RTP traffic, I have softphone connecting
>> from the outside, I can register but Kamailio is sending all RTP traffic
>> to the private IP address that the softphone has in it own local
>> network. How can I fix this? Any help will be appreciated.
>>
>
> That is what the far-end NAT traversal / rtpproxy part of Kamailio
> functionality is for.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.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
>
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From miconda at gmail.com  Thu Jun  2 08:10:30 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 02 Jun 2011 08:10:30 +0200
Subject: [SR-Users] [sr-dev] discussing about next major release - v3.2.0
In-Reply-To: <4DE4B29F.9000403@gmail.com>
References: <4DD623E0.9020807@gmail.com> <4DD62834.1020000@gmail.com>
	<BANLkTikZXyStD18vDHok=i-AfKnSDS9Yaw@mail.gmail.com>
	<201105201647.06894.henning.westerholt@1und1.de>
	<4DE4B29F.9000403@gmail.com>
Message-ID: <4DE72956.9060809@gmail.com>

So latest updates render several people unavailable today, so tomorrow, 
Friday, June 3, 15:00GMT is the winner. Let's lock it down.

See you on IRC freenode.net, channeL #sip-router at 15:00GMT.

Cheers,
Daniel

On 5/31/11 11:19 AM, Daniel-Constantin Mierla wrote:
> Based on people that expressed the option, we can do it Thursday this 
> week, is that fine? If not Friday.
>
> If anyone wants to announce himself/herself, here is again the link.
> http://doodle.com/sipgepeyp4us4yey
>
> Cheers,
> Daniel
>
> On 5/20/11 4:47 PM, Henning Westerholt wrote:
>> On Friday 20 May 2011, Carsten Bock wrote:
>>> here is the doodle link:
>>> http://doodle.com/sipgepeyp4us4yey
>>>
>>> If anyone suggests other dates, i can add them accordingly.
>> Hi Carsten,
>>
>> great, i've added my availability as well.
>>
>> Cheers,
>>
>> Henning
>>
>> _______________________________________________
>> 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



From ozren.lapcevic at gmail.com  Thu Jun  2 15:57:07 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Thu, 2 Jun 2011 15:57:07 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
In-Reply-To: <4DE64CBE.3030607@gmail.com>
References: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
	<4DE64CBE.3030607@gmail.com>
Message-ID: <BANLkTinDze_Kk4rBbrtwEfy_Ub8WKWFDag@mail.gmail.com>

Hi Daniel,

here is the backtrace:

#0  0x00007ffc46b81ed5 in raise () from /lib/libc.so.6
#1  0x00007ffc46b833f3 in abort () from /lib/libc.so.6
#2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
main.c:651
#3  <signal handler called>
#4  0x00007ffc46c1c307 in syscall () from /lib/libc.so.6
#5  0x00007ffc4219f115 in imc_del_room (name=0x7ffc3fe0d388,
domain=0x7ffc3fe0d398) at ../../mem/../futexlock.h:110
#6  0x00007ffc4219f1d8 in imc_htable_destroy () at imc_mng.c:104
#7  0x00007ffc4219347a in destroy () at imc.c:736
#8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
#9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
#10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
show_status=1) at main.c:668
#11 0x0000000000463e9c in handle_sigs () at main.c:759
#12 0x0000000000465572 in main_loop () at main.c:1676
#13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
main.c:2398

I'm not sure if I got the right thing. I got it with gdb
/usr/local/kamailio-ozren/sbin/kamailio --core core and called bt in gdb.

I've repeated scenario described in the first mail: user tries to join in a
private room without getting an invite first.

Best regards,
Ozren


On Wed, Jun 1, 2011 at 4:29 PM, Daniel-Constantin Mierla
<miconda at gmail.com>wrote:

>  Hello,
>
> can you get the core dump and send the backtrace?
>
> Be sure you have 'ulimit -c unlimited', working directory (-w option) set
> to writable path and you don't run it in chowner (no -u or -g parameters).
>
> Cheers,
> Daniel
>
> On 6/1/11 4:25 PM, Ozren Lapcevic wrote:
>
> Hi,
>
> there is a bug in IMC module that crashes Kamailio.
>
> Here is the scenario: One user (pero at abc.hr) creates private chat room (
> chat-999 at abc.hr). Other user (oz at abc.hr) tries to join the chat-999 at abc.hrby sending #join. Kamailio crashes at that point. Here is the log:
>
> Best regards,
> Ozren
>
>
> Jun  1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]: segfault
> at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in imc.so[7f28f89cf000+17000]
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-999 at abc.hr] B
> [#join]
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: <core> [receive.c:186]: preparing to run routing scripts...
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
> DEBUG: imc [imc.c:525]: found command
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: maxfwd [mf_funcs.c:85]: value = 70
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
> DEBUG: imc [imc_cmd.c:146]: command: [join]
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body :
> content_length=223
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
> DEBUG: imc [imc_cmd.c:292]: found room [chat-999]
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: <core> [parser/msg_parser.c:103]: found end of header
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
> ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user
> [oz]
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
> ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
> DEBUG: <core> [parser/msg_parser.c:630]: SIP Request:
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
> ALERT: <core> [main.c:744]: core was not generated
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
> DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82,
> 161.53.255.82, 0)
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
> DEBUG: <core> [parser/msg_parser.c:632]:  method:  <MESSAGE>
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
> INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]:
> INFO: <core> [main.c:807]: INFO: signal 15 received
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
> DEBUG: imc [imc.c:709]: room 3 chat-777
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
> DEBUG: imc [imc.c:709]: room 11 chat-888
> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
> DEBUG: imc [imc.c:709]: room 13 chat-999
>
>
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From ozren.lapcevic at gmail.com  Thu Jun  2 16:54:26 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Thu, 2 Jun 2011 16:54:26 +0200
Subject: [SR-Users] IMC module unresolved issues
In-Reply-To: <4DE64A4C.7090201@gmail.com>
References: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>
	<4DE6173D.6000409@gmail.com>
	<BANLkTikmtRezEkXE+wAz1oYxSV=FXGR65w@mail.gmail.com>
	<4DE64A4C.7090201@gmail.com>
Message-ID: <BANLkTimUq4kJqqRsDcLQT733xRDdMasNoA@mail.gmail.com>

On Wed, Jun 1, 2011 at 4:18 PM, Daniel-Constantin Mierla
<miconda at gmail.com>wrote:

>
> Any issues still going on with the module?
>
> Best regards,
> Daniel
>


Hi Daniel,

yes, there are still several issues we've found. I hope that you or the
other authors of the module will find some time to look into the issues.

1. last member in the chat room exits + call of ./kamctl fifo imc_list_rooms
crashes Kamailio.
Working example (no bugs): oz at abc.hr creates chat room chat-555 at abc.hr.
Doesn't write messages to the chat room. Exits (#exit). Receives message:
"The room has been destroyed". Call of ./kamctl fifo imc_list_rooms works
fine and doesn't display the room which was just destroyed.

Example which crashes Kamailio: oz at abc.hr creates chat room chat-555 at abc.hr.
Writes something to the chat room. Exits (#exit). No messages are displayed
in SIP client, although he should get "The room has been destroyed". Call of
./kamctl fifo imc_list_rooms doesn't display anything (although there are
other rooms besides chat-555 at abc.hr) and crashes Kamailio. In the database,
in imc_rooms, chat-555 at abc.hr is still there, but there are no members in
imc_members table. Log:
#processing of #exit

Jun  2 16:21:49 ser-test ./kamailio[5547]: INFO: <script>: going to imc: F [
sip:oz at abc.hr] T [sip:chat-555 at abc.hr] B [#exit]
Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc.c:525]: found
command
Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc_cmd.c:146]:
command: [exit]
Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc_mng.c:373]: found
member
Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc_cmd.c:1091]: nr =
0

#Call of ./kamctl fifo imc_list_rooms

Jun  2 16:22:02 ser-test ./kamailio[5550]: DEBUG: mi_fifo [fifo_fnc.c:493]:
entered consume
Jun  2 16:22:02 ser-test kernel: [5934996.535094] kamailio[5550]: segfault
at 10 ip 7f85c0b92a46 sp 7fffffffc700 error 4 in imc.so[7f85c0b8c000+17000]
Jun  2 16:22:02 ser-test ./kamailio[5550]: DEBUG: mi_fifo [fifo_fnc.c:493]:
**** done consume
Jun  2 16:22:02 ser-test ./kamailio[5550]: DEBUG: mi_fifo [fifo_fnc.c:507]:
done parsing the mi tree
Jun  2 16:22:02 ser-test ./kamailio[5557]: : <core> [pass_fd.c:293]: ERROR:
receive_fd: EOF on 23
Jun  2 16:22:02 ser-test ./kamailio[5557]: DEBUG: <core> [tcp_main.c:3554]:
DBG: handle_ser_child: dead child 11, pid 5550 (shutting down?)
Jun  2 16:22:02 ser-test ./kamailio[5557]: DEBUG: <core> [io_wait.h:617]:
DBG: io_watch_del (0x81b400, 23, -1, 0x0) fd_no=23 called
Jun  2 16:22:02 ser-test ./kamailio[5539]: ALERT: <core> [main.c:741]: child
process 5550 exited by a signal 11
Jun  2 16:22:02 ser-test ./kamailio[5539]: ALERT: <core> [main.c:744]: core
was generated
Jun  2 16:22:02 ser-test ./kamailio[5539]: INFO: <core> [main.c:756]: INFO:
terminating due to SIGCHLD
Jun  2 16:22:02 ser-test ./kamailio[5540]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5541]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5542]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5543]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5544]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5545]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5546]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5547]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5548]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5551]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5552]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5553]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5549]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5554]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5555]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5556]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5557]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  2 16:22:02 ser-test ./kamailio[5539]: DEBUG: imc [imc.c:709]: room 11
chat-555

Backtrace: (same as for #join to private conference described in other
topic!)
#0  0x00007f85c557eed5 in raise () from /lib/libc.so.6
#1  0x00007f85c55803f3 in abort () from /lib/libc.so.6
#2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
main.c:651
#3  <signal handler called>
#4  0x00007f85c5619307 in syscall () from /lib/libc.so.6
#5  0x00007f85c0b9c0c6 in imc_del_room (name=0x7f85be80a388,
domain=0x7f85be80a398) at ../../mem/../futexlock.h:123
#6  0x00007f85c0b9c1d8 in imc_htable_destroy () at imc_mng.c:104
#7  0x00007f85c0b9047a in destroy () at imc.c:736
#8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
#9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
#10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
show_status=1) at main.c:668
#11 0x0000000000463e9c in handle_sigs () at main.c:759
#12 0x0000000000465572 in main_loop () at main.c:1676
#13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
main.c:2398


2. #remove command prints wrong info.
Example: oz at abc.hr creates chat room chat-555 at abc.hr. Invites
slavko at abc.hr(#invite
slavko at abc.hr). Invites pero at abc.hr. Both of them join with #accept.
oz at abc.hr removes pero (#remove pero at abc.hr). Pero's client displays
messages: "You have been removed from this room." and Pero can't send
messages to chat room any more. However, Oz's and Slavko's clients display
message: "<sip:pero at abc.hr> has joined the room".


3. #invite & #remove commands do not work after typing some text in
conference room.
Example 1: oz at abc.hr creates chat room chat-555 at abc.hr. Invites
slavko at abc.hr (#invite slavko at abc.hr). Invites pero at abc.hr. Both of them
join with #accept. Pero and Slavko write messages to chat room. Oz can
remove them at this point. However, after Oz writes a message to chat room,
he can't remove members any more and receives 500 command error. Log:

Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-555 at abc.hr] B
[#remove slavko at abc.hr]
Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
DEBUG: imc [imc.c:525]: found command
Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
DEBUG: imc [imc_cmd.c:146]: command: [remove]
Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
DEBUG: imc [imc_cmd.c:151]: parameter 0=[slavko at abc.hr]
Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
DEBUG: imc [imc_mng.c:373]: found member
Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
ERROR: imc [imc_cmd.c:664]: user [oz] has no right to remove other users [
sip:slavko at abc.hr]!
Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
ERROR: imc [imc.c:578]: failed to handle 'remove'

./kamctl fifo imc_list_rooms
ROOM::  URI=sip:chat-555 at abc.hr MEMBERS=0 OWNER=sip:oz at abc.hr

./kamctl fifo imc_list_members sip:chat-555 at abc.hr
ROOM:: sip:chat-555 at abc.hr NR_OF_MEMBERS=2
        MEMBER:: sip:oz at abc.hr
        MEMBER:: sip:slavko at abc.hr

Example 2: oz at abc.hr creates chat room chat-666 at abc.hr. Invites
slavko at abc.hr. Slavko joins with #accept. Oz writes a message to chat room.
Invites pero at abc.hr. Imc manager returns 500 command error. (NOTE: If Oz
doesn't write any text to chat room, he can invite multiple users.) Here is
the log:

Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-666 at abc.hr] B
[#invite pero at abc.hr]
Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
DEBUG: imc [imc.c:525]: found command
Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
DEBUG: imc [imc_cmd.c:146]: command: [invite]
Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
DEBUG: imc [imc_cmd.c:151]: parameter 0=[pero at abc.hr]
Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
DEBUG: imc [imc_mng.c:373]: found member
Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
ERROR: imc [imc_cmd.c:441]: user [oz] has no right to invite other users!
Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
ERROR: imc [imc.c:554]: failed to handle 'invite'

./kamctl fifo imc_list_rooms
ROOM::  URI=sip:chat-666 at abc.hr MEMBERS=0 OWNER=sip:oz at abc.hr

./kamctl fifo imc_list_members sip:chat-666 at abc.hr
ROOM:: sip:chat-666 at abc.hr NR_OF_MEMBERS=2
        MEMBER:: sip:oz at abc.hr
        MEMBER:: sip:slavko at abc.hr


4. ./kamctl fifo imc_list_rooms command doesn't report correct number of
members in the room, number of members remains 0, although members have
successfully joined the chat room and ./kamctl fifo imc_list_members
sip:chat-555 at abc.hr returns correct number of members.


Best regards,
Ozren
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From agranig at sipwise.com  Thu Jun  2 21:23:47 2011
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 02 Jun 2011 21:23:47 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
Message-ID: <4DE7E343.1090509@sipwise.com>

Hi,

Found an interesting call-flow today where a client (pjsip) is behind
NAT and behaves like this during registration (kamailio-3.1.3):

REGISTER sip:somedomain.com
Contact: <sip:me at 10.0.0.7:2001;ob>
Expires: 900

SIP/2.0 200 OK
Contact: <sip:me at 10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001"

REGISTER sip:somedomain.com
Contact: <sip:me at 1.2.3.4:3001;transport=UDP;ob>
Contact: <sip:me at 10.0.0.7:2001;ob>;expires=0
Expires: 900

SIP/2.0 200 OK
Contact:
<sip:me at 10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001",
<sip:me at 1.2.3.4:3001;transport=UDP;ob>;expires=900

So obviously the client is trying to unregister its private contact (for
whatever reason - probably to play nice with registrars not supporting
far-end NAT traversal), and tries to register the Contact with the
"received" address instead. Kamailio however ignores the ";expires=0"
for the private Contact and just saves both of them with the expiry
value given in the Expires header.

Taking the client behavior aside for now: is this the correct behavior
of the registrar module? In RFC3261 chapter 10.2.1.1 it says:

#+
There are two ways in which a client can suggest an expiration interval
for a binding: through an Expires header field or an "expires" Contact
header parameter. The latter allows expiration intervals to be suggested
on a per-binding basis when more than one binding is given in a single
REGISTER request, whereas the former suggests an expiration interval for
all Contact header field values that do not contain the "expires"
parameter.
#-

So I'd say that in my specific case "expires" Contact header param
(which is 0 for the binding having 10.0.0.7) should de-register this
particular binding, whereas the Expires header should be applied to the
second binding (having the 1.2.3.4) because it doesn't carry an
"expires" param.

Opinions on that one?

Andreas

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From gchen00 at insightbb.com  Thu Jun  2 21:26:13 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Thu, 2 Jun 2011 15:26:13 -0400 (EDT)
Subject: [SR-Users] Problem using dispatcher.
In-Reply-To: <812663188.252560.1307040530289.JavaMail.root@md09.insight.synacor.com>
Message-ID: <488009766.254051.1307042773299.JavaMail.root@md09.insight.synacor.com>

Kamailio 3.1.2 
Here is my setup: 

Phone1 (10.10.1.114) 
| 
Kamailio (10.10.1.230) 
/ \ 
/ \ 
Freeswitch1 Freeswitch2 ----------------------> Other phones 
(10.10.1.222) (10.10.1.223) 


Both Freeswitch's share the same REGISTRATION information. Phone1 is already registered with Freeswitch. When I try to dial 'Other phones' using phone1, the dispatcher always get into FAILURE route although the log shows that kamailio did send INVITE to 10.10.1.222 and received '407 - Proxy Authentication Required' from Freexwitch1 and then dispatcher send INVITE to 10.10.1.223 and also received '407 - Proxy Authentication Required' from Freexwitch2 and eventually failed. 


Any idea? 


Here is my configuration: 

modparam("dispatcher", "db_url", DBURL) 
modparam("dispatcher", "table_name", "dispatcher") 
modparam("dispatcher", "ds_ping_interval", 10000) 
modparam("dispatcher", "ds_probing_threshhold", 3) 
#modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") 
modparam("dispatcher", "ds_ping_method", "OPTIONS") 
modparam("dispatcher", "ds_probing_mode", 1) 
modparam("dispatcher", "ds_ping_from", "sip:proxy at SIP-router1.testnet.net") 
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", "flags", 3) 
modparam("dispatcher", "force_dst", 1) 


route { 
.... 

if ( ds_select_dst("$var(group)", "9") ){ 
route(RELAY_DS); 
return; 
... 
} 

route[[RELAY_DS] 
if (is_method("INVITE")) { 
t_on_branch("BRANCH_ONE"); 
t_on_reply("NAT_REPLY"); 
t_on_failure("FAIL_DS"); 
} 
if (!t_relay()) { 
sl_reply_error(); 
} 
} 

failure_route[FAIL_DS] { 
if (t_check_status("408")) { 
ds_mark_dst("i"); 
} 
if (!ds_next_dst()) { 
t_reply("503", "Service unavailable: no more dst"); 
exit; 
} 
t_on_failure("FAIL_DS"); 
route(RELAY_DS); 
} 

onreply_route[NAT_REPLY] 
{ 
xlog("L_INFO", "REPLY_ROUTE: S=$rs - $rr F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n"); 

} 


And here is the SIP messages from ngrep: 

U 10.10.1.144:51029 -> 10.10.1.230:5060 
INVITE sip:1001 at fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..F 
rom: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet 
.net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..Max-Forwards: 70..Date: Thu, 02 Jun 2011 
19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011 at 10.10.1.144:5060>..Expires: 
180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..C 
ontent-Length: 261..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 
10962 0 IN IP4 10.10.1.144..s=SIP Call..t=0 0..m=audio 20372 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.144..a=rtpmap: 
0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv 
.. 
# 
U 10.10.1.230:5060 -> 10.10.1.144:5060 
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: 
"Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net; 
user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x8 
6_64/linux))..Content-Length: 0.... 
# 
U 10.10.1.230:5060 -> 10.10.1.222:5060 
INVITE sip:1001 at fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..Via: SIP/2.0/ 
UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "L 
ine1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net;us 
er=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 2011 19:14: 
14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011 at 10.10.1.144:5060>..Expires: 180..A 
ccept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..Content- 
Length: 277..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 
IN IP4 10.10.1.231..s=SIP Call..t=0 0..m=audio 38426 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.231..a=rtpmap:0 PCMU/80 
00..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=nortp 
proxy:yes.. 
# 
U 10.10.1.222:5060 -> 10.10.1.230:5060 
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060 
;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..From: "Line1" <sip:5022441011 at fs2000.lightyea 
r.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net;user=phone>..Call-ID: 00082166-efcb0013 
-692f44d9-6760a891 at 10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 0 
3-05-32 -0400..Content-Length: 0.... 
# 
U 10.10.1.222:5060 -> 10.10.1.230:5060 
SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/U 
DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb0021 
53bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9 
-6760a891 at 10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 
0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, 
PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se 
ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea 
lm="fs2000.testnet.net", nonce="8da1ddb1-e393-442c-8be0-17e439457363", algorithm=MD5, qop="auth"..Content-Length: 0... 
. 
# 
U 10.10.1.230:5060 -> 10.10.1.222:5060 
ACK sip:1001 at fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Fro 
m: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.n 
et;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..Max-Forwards: 69..CSeq: 
101 ACK..Content-Length: 0.... 
# 
U 10.10.1.230:5060 -> 10.10.1.223:5060 
INVITE sip:1001 at fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..Via: 
SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d.. 
From: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.lightyea 
r.net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 201 
1 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011 at 10.10.1.144:5060>..Expires 
: 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces.. 
Content-Length: 328..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 
10962 0 IN IP4 10.10.1.23110.10.1.231..s=SIP Call..t=0 0..m=audio 3842638426 RTP/AVP 0 8 18 101..c=IN IP4 216.49. 
129.23110.10.1.231..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8 
000..a=fmtp:101 0-15..a=sendrecv..a=nortpproxy:yes..a=nortpproxy:yes.. 
# 
U 10.10.1.223:5060 -> 10.10.1.230:5060 
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060 
;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..From: "Line1" <sip:5022441011 at fs2000. 
testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net;user=phone>..Call-ID: 00082166- 
efcb0013-692f44d9-6760a891 at 10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011 
-04-19 03-05-32 -0400..Content-Length: 0.... 
# 
U 10.10.1.223:5060 -> 10.10.1.230:5060 
SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/U 
DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb0021 
53bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9 
-6760a891 at 10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 
0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, 
PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se 
ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea 
lm="fs2000.testnet.net", nonce="8769298b-c52d-43b8-b69c-6ebf26f3184d", algorithm=MD5, qop="auth"..Content-Length: 0... 
. 
# 
U 10.10.1.230:5060 -> 10.10.1.223:5060 
ACK sip:1001 at fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Fro m: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.n 
et;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..Max-Forwards: 69..CSeq: 
101 ACK..Content-Length: 0.... 
# 
U 10.10.1.230:5060 -> 10.10.1.144:5060 
SIP/2.0 503 Service unavailable: no more dst..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1 
" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.net;user=p 
hone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..CSeq: 101 
INVITE..Server: kamailio (3.1.2 (x86_64/linux))..Content-Length: 0.... 
# 
U 10.10.1.144:51030 -> 10.10.1.230:5060 
ACK sip:1001 at fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From 
: "Line1" <sip:5022441011 at fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001 at fs2000.testnet.ne 
t;user=phone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891 at 10.10.1.144..D 
ate: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 ACK..Content-Length: 0.... 




Gary Chen 
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From abalashov at evaristesys.com  Thu Jun  2 21:30:14 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 02 Jun 2011 15:30:14 -0400
Subject: [SR-Users] Problem using dispatcher.
In-Reply-To: <488009766.254051.1307042773299.JavaMail.root@md09.insight.synacor.com>
References: <488009766.254051.1307042773299.JavaMail.root@md09.insight.synacor.com>
Message-ID: <4DE7E4C6.7030301@evaristesys.com>

On 06/02/2011 03:26 PM, Gary Chen wrote:

> the dispatcher always get into FAILURE route although the log shows
> that kamailio did send INVITE to 10.10.1.222 and received '407 -
> Proxy Authentication Required' from Freexwitch1 and then dispatcher
> send INVITE to 10.10.1.223 and also received '407 - Proxy
> Authentication Required' from Freexwitch2 and eventually failed.

That's because any reply >= 300 is considered a failure.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From gchen00 at insightbb.com  Thu Jun  2 21:41:14 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Thu, 2 Jun 2011 15:41:14 -0400 (EDT)
Subject: [SR-Users] Problem using dispatcher.
In-Reply-To: <252320405.254378.1307043192332.JavaMail.root@md09.insight.synacor.com>
Message-ID: <162648092.254595.1307043674889.JavaMail.root@md09.insight.synacor.com>



Thanks for the quick reply. 

How can I fix this? 

We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? 

I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. 




Any suggestion? 



-----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: Thursday, June 02, 2011 3:30 PM 
To: sr-users at lists.sip-router.org 
Subject: Re: [SR-Users] Problem using dispatcher. 



On 06/02/2011 03:26 PM, Gary Chen wrote: 



> the dispatcher always get into FAILURE route although the log shows 

> that kamailio did send INVITE to 10.10.1.222 and received '407 - 

> Proxy Authentication Required' from Freexwitch1 and then dispatcher 

> send INVITE to 10.10.1.223 and also received '407 - Proxy 

> Authentication Required' from Freexwitch2 and eventually failed. 



That's because any reply >= 300 is considered a failure. 



-- 

Alex Balashov - Principal 

Evariste Systems LLC 

260 Peachtree Street NW 

Suite 2200 

Atlanta, GA 30303 

Tel: +1-678-954-0670 

Fax: +1-404-961-1892 

Web: http://www.evaristesys.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 
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From miconda at gmail.com  Thu Jun  2 22:24:28 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 02 Jun 2011 22:24:28 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
In-Reply-To: <4DE7E343.1090509@sipwise.com>
References: <4DE7E343.1090509@sipwise.com>
Message-ID: <4DE7F17C.7070709@gmail.com>

Hello,

I looked in the code and the expires is taken from contact header 
parameter if present. Maybe you can run in higher debug level and see if 
you can spot something in the log messages that can give a lead.

Cheers,
Daniel

On 6/2/11 9:23 PM, Andreas Granig wrote:
> Hi,
>
> Found an interesting call-flow today where a client (pjsip) is behind
> NAT and behaves like this during registration (kamailio-3.1.3):
>
> REGISTER sip:somedomain.com
> Contact:<sip:me at 10.0.0.7:2001;ob>
> Expires: 900
>
> SIP/2.0 200 OK
> Contact:<sip:me at 10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001"
>
> REGISTER sip:somedomain.com
> Contact:<sip:me at 1.2.3.4:3001;transport=UDP;ob>
> Contact:<sip:me at 10.0.0.7:2001;ob>;expires=0
> Expires: 900
>
> SIP/2.0 200 OK
> Contact:
> <sip:me at 10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001",
> <sip:me at 1.2.3.4:3001;transport=UDP;ob>;expires=900
>
> So obviously the client is trying to unregister its private contact (for
> whatever reason - probably to play nice with registrars not supporting
> far-end NAT traversal), and tries to register the Contact with the
> "received" address instead. Kamailio however ignores the ";expires=0"
> for the private Contact and just saves both of them with the expiry
> value given in the Expires header.
>
> Taking the client behavior aside for now: is this the correct behavior
> of the registrar module? In RFC3261 chapter 10.2.1.1 it says:
>
> #+
> There are two ways in which a client can suggest an expiration interval
> for a binding: through an Expires header field or an "expires" Contact
> header parameter. The latter allows expiration intervals to be suggested
> on a per-binding basis when more than one binding is given in a single
> REGISTER request, whereas the former suggests an expiration interval for
> all Contact header field values that do not contain the "expires"
> parameter.
> #-
>
> So I'd say that in my specific case "expires" Contact header param
> (which is 0 for the binding having 10.0.0.7) should de-register this
> particular binding, whereas the Expires header should be applied to the
> second binding (having the 1.2.3.4) because it doesn't carry an
> "expires" param.
>
> Opinions on that one?
>
> 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

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From miconda at gmail.com  Thu Jun  2 22:28:16 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 02 Jun 2011 22:28:16 +0200
Subject: [SR-Users] Problem using dispatcher.
In-Reply-To: <162648092.254595.1307043674889.JavaMail.root@md09.insight.synacor.com>
References: <162648092.254595.1307043674889.JavaMail.root@md09.insight.synacor.com>
Message-ID: <4DE7F260.2050701@gmail.com>



On 6/2/11 9:41 PM, Gary Chen wrote:
>
>  Thanks for the quick reply.
>
> How can I fix this?
>
When you set t_on_failure for an INVITE, the respective failure route 
will be executed for any reply >=0. What you have to do is to not 
re-route to a new destination from failure route, simply exit for those 
reply codes, like:

if(t_check_status("401|407")) exit;

You don't handle as well the canceled transactions, when re-routing 
makes no sense -- you should do it (see default config file for example.

Cheers,
Daniel

> We are using Kamailio as single point of entry for all our phones. Our 
> phones all registered with Freeswitch. Kamailio only route the SIP 
> traffic. Is there any way I can make Kamailio not treat 407 as failure?
>
> I also have the same problem with REGISTER messages when Freeswitch 
> reply with '401 Unauthorized'. I end up using Forward() instead of 
> t_relay() to make the phone1 register with Freeswitch.
>
>
> Any suggestion?
>
> -----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: Thursday, June 02, 2011 3:30 PM
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] Problem using dispatcher.
>
> On 06/02/2011 03:26 PM, Gary Chen wrote:
>
> > the dispatcher always get into FAILURE route although the log shows
>
> > that kamailio did send INVITE to 10.10.1.222 and received '407 -
>
> > Proxy Authentication Required' from Freexwitch1 and then dispatcher
>
> > send INVITE to 10.10.1.223 and also received '407 - Proxy
>
> > Authentication Required' from Freexwitch2 and eventually failed.
>
> That's because any reply >= 300 is considered a failure.
>
> -- 
>
> Alex Balashov - Principal
>
> Evariste Systems LLC
>
> 260 Peachtree Street NW
>
> Suite 2200
>
> Atlanta, GA 30303
>
> Tel: +1-678-954-0670
>
> Fax: +1-404-961-1892
>
> Web: http://www.evaristesys.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
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From agranig at sipwise.com  Thu Jun  2 22:49:07 2011
From: agranig at sipwise.com (Andreas Granig)
Date: Thu, 02 Jun 2011 22:49:07 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
In-Reply-To: <4DE7F17C.7070709@gmail.com>
References: <4DE7E343.1090509@sipwise.com> <4DE7F17C.7070709@gmail.com>
Message-ID: <4DE7F743.4090501@sipwise.com>

Hi Daniel,

On 06/02/2011 10:24 PM, Daniel-Constantin Mierla wrote:
> Hello,
> 
> I looked in the code and the expires is taken from contact header
> parameter if present. Maybe you can run in higher debug level and see if
> you can spot something in the log messages that can give a lead.

Well, the only thing which looks interesting is this:

DEBUG: registrar [save.c:496]: 1 valid contacts
DEBUG: registrar [save.c:518]: 1 contacts after commit

And later:

DEBUG: registrar [reply.c:192]: created Contact HF: Contact:
<sip:me at 10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001",
<sip:me at 1.2.3.4:3001;transport=UDP;ob>;expires=900

So probably it just recognizes the second Contact (the one with the
public IP and without the expires param)?

Andreas

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From gchen00 at insightbb.com  Thu Jun  2 23:07:09 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Thu, 2 Jun 2011 17:07:09 -0400 (EDT)
Subject: [SR-Users] Problem using dispatcher.
Message-ID: <783560745.257168.1307048829494.JavaMail.root@md09.insight.synacor.com>



That works. Thanks. 





From: Daniel-Constantin Mierla [mailto:miconda at gmail.com] 
Sent: Thursday, June 02, 2011 4:28 PM 
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List 
Cc: Gary Chen 
Subject: Re: [SR-Users] Problem using dispatcher. 





On 6/2/11 9:41 PM, Gary Chen wrote: 

Thanks for the quick reply. 

How can I fix this? 

When you set t_on_failure for an INVITE, the respective failure route will be executed for any reply >=0. What you have to do is to not re-route to a new destination from failure route, simply exit for those reply codes, like: 

if(t_check_status("401|407")) exit; 

You don't handle as well the canceled transactions, when re-routing makes no sense -- you should do it (see default config file for example. 

Cheers, 
Daniel 




We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? 

I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. 



Any suggestion? 



-----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: Thursday, June 02, 2011 3:30 PM 
To: sr-users at lists.sip-router.org 
Subject: Re: [SR-Users] Problem using dispatcher. 



On 06/02/2011 03:26 PM, Gary Chen wrote: 



> the dispatcher always get into FAILURE route although the log shows 

> that kamailio did send INVITE to 10.10.1.222 and received '407 - 

> Proxy Authentication Required' from Freexwitch1 and then dispatcher 

> send INVITE to 10.10.1.223 and also received '407 - Proxy 

> Authentication Required' from Freexwitch2 and eventually failed. 



That's because any reply >= 300 is considered a failure. 



-- 

Alex Balashov - Principal 

Evariste Systems LLC 

260 Peachtree Street NW 

Suite 2200 

Atlanta, GA 30303 

Tel: +1-678-954-0670 

Fax: +1-404-961-1892 

Web: http://www.evaristesys.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 






_______________________________________________ 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://linkedin.com/in/miconda -- http://twitter.com/miconda 
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From bruno.bresciani at gmail.com  Fri Jun  3 00:07:47 2011
From: bruno.bresciani at gmail.com (Bruno Bresciani)
Date: Thu, 2 Jun 2011 19:07:47 -0300
Subject: [SR-Users] callback function of tm module
Message-ID: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>

Hi All,

I'm having doubts in the implementation of the callback function module tm.
As she calls the function to record register_tmcb ()?
Can someome help me?


Regards
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From miconda at gmail.com  Fri Jun  3 00:10:11 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 00:10:11 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
In-Reply-To: <4DE7F743.4090501@sipwise.com>
References: <4DE7E343.1090509@sipwise.com> <4DE7F17C.7070709@gmail.com>
	<4DE7F743.4090501@sipwise.com>
Message-ID: <4DE80A43.5000706@gmail.com>

Hello,

one more thing, what is the output of kamctl ul show? I want to see if 
it is a matter of building the contact header for reply or is in 
internal list of location records.

Cheers,
Daniel

On 6/2/11 10:49 PM, Andreas Granig wrote:
> Hi Daniel,
>
> On 06/02/2011 10:24 PM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> I looked in the code and the expires is taken from contact header
>> parameter if present. Maybe you can run in higher debug level and see if
>> you can spot something in the log messages that can give a lead.
> Well, the only thing which looks interesting is this:
>
> DEBUG: registrar [save.c:496]: 1 valid contacts
> DEBUG: registrar [save.c:518]: 1 contacts after commit
>
> And later:
>
> DEBUG: registrar [reply.c:192]: created Contact HF: Contact:
> <sip:me at 10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001",
> <sip:me at 1.2.3.4:3001;transport=UDP;ob>;expires=900
>
> So probably it just recognizes the second Contact (the one with the
> public IP and without the expires param)?
>
> Andreas

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From agranig at sipwise.com  Fri Jun  3 00:16:33 2011
From: agranig at sipwise.com (Andreas Granig)
Date: Fri, 03 Jun 2011 00:16:33 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
In-Reply-To: <4DE80A43.5000706@gmail.com>
References: <4DE7E343.1090509@sipwise.com> <4DE7F17C.7070709@gmail.com>
	<4DE7F743.4090501@sipwise.com> <4DE80A43.5000706@gmail.com>
Message-ID: <4DE80BC1.3090903@sipwise.com>

Hi,

On 06/03/2011 12:10 AM, Daniel-Constantin Mierla wrote:
> one more thing, what is the output of kamctl ul show? I want to see if
> it is a matter of building the contact header for reply or is in
> internal list of location records.

Domain:: location table=512 records=1 max_slot=1
	AOR:: me at somedomain.com
		Contact:: sip:me at 10.0.0.7:52030;ob Q=
			Expires:: 875
			Callid:: DZZHTDttFd5WivydjNnWfpRLsXp8RIYh
			Cseq:: 4599
			User-agent:: myclient
			Received:: sip:1.2.3.4:55642
			Path:: <sip:lb at 127.0.0.1:5060;lr;received=sip:1.2.3.4:55642>
			State:: CS_SYNC
			Flags:: 0
			Cflag:: 192
			Socket:: udp:127.0.0.1:5062
			Methods:: 8143
		Contact:: sip:me at 1.2.3.4:55642;transport=UDP;ob Q=
			Expires:: 876
			Callid:: DZZHTDttFd5WivydjNnWfpRLsXp8RIYh
			Cseq:: 4601
			User-agent:: myclient
			Path:: <sip:lb at 127.0.0.1:5060;lr>
			State:: CS_SYNC
			Flags:: 0
			Cflag:: 0
			Socket:: udp:127.0.0.1:5062
			Methods:: 8143

Hope this helps,
Andreas

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From mokhtar.bengana at gmail.com  Fri Jun  3 03:45:47 2011
From: mokhtar.bengana at gmail.com (Mokhtar Bengana)
Date: Thu, 2 Jun 2011 20:45:47 -0500
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
Message-ID: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>

I have Kamailio and rtpproxy running behind NAT on the same server.
Inbound calls to Kamailio from the outside work fine and I have two
way audio but when I try an outbound echo test call the call connects
but I have no audio. rtpproxy seems to be running and all rtp ports
are poiting to Kamailio internal ip. The syslog is showing this error.
Not sure what's missing in the config. Appreciate any inputs. Thanks.



root at Kamailio:~# tail /var/log/syslog
Jun  3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy
[rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:22222> found, support for
it enabled
Jun  3 01:25:05 Kamailio kamailio[791]: INFO: <core> [forward.c:786]:
broken reply to forward - no 2nd via
Jun  3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy
[rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
Jun  3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy
[rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy


From miconda at gmail.com  Fri Jun  3 09:15:03 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 09:15:03 +0200
Subject: [SR-Users] IRC meeting today, 15:00GMT
Message-ID: <4DE889F7.5080505@gmail.com>

Hello,

everyone that wants to discuss about the next major release (v3.2.0), 
please join us on IRC:

Channel: #sip-router
Server: irc.freenode.net
Time: 15:00GMT/London (16:00 Berlin/Paris/Madrid, 10:00am New York)

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Fri Jun  3 09:16:42 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 09:16:42 +0200
Subject: [SR-Users] perlvdb error
In-Reply-To: <4DD4CA9B.3040109@gmail.com>
References: <BANLkTik+3HUgwA3HGfhfJRNVK_bJycccmw@mail.gmail.com>
	<4DAC09CA.7080206@gmail.com>
	<BANLkTi=BKYYRu1KpfU73kCG21iwnrM==Bg@mail.gmail.com>
	<201104181447.03413.henning.westerholt@1und1.de>
	<BANLkTinrfwujAZSdwS+iBQxStaj5OX7uxg@mail.gmail.com>
	<BANLkTi=NpFr5MRBLnVQ0TfbmB35ELB_VBQ@mail.gmail.com>
	<BANLkTinxK-nj_NY+nszFQigN_txgwWjWkg@mail.gmail.com>
	<4DAED065.2040300@gmail.com>
	<BANLkTinoqO5g=FwngV4CEmo5CJnGQ8Y1xg@mail.gmail.com>
	<4DAFF5F4.1020805@gmail.com>
	<BANLkTin8YzyNqrZ9sHDLsMrK4MATT1tTeA@mail.gmail.com>
	<4DB6891E.4060102@gmail.com>
	<BANLkTimX=bjhP1pgZ+WPo4SBVFBqo2kU1Q@mail.gmail.com>
	<4DB82787.6020401@gmail.com>
	<BANLkTinkxHA2u69WRgafMKL=6=EZHGVCNA@mail.gmail.com>
	<BANLkTinLif8+Gh6hM5MzzrQD6qxRGLWxvQ@mail.gmail.com>
	<4DBEA7DD.7070500@gmail.com>
	<BANLkTikj4zZcqNgVzPhyj9AbZ5B9sy7jsA@mail.gmail.com>
	<4DD4CA9B.3040109@gmail.com>
Message-ID: <4DE88A5A.2010206@gmail.com>



On 5/19/11 9:45 AM, Daniel-Constantin Mierla wrote:
>
>
> On 5/18/11 8:29 AM, Roman Yeryomin wrote:
>> On 2 May 2011 15:47, Daniel-Constantin Mierla<miconda at gmail.com>  wrote:
>>>
>>> On 5/1/11 11:51 PM, Roman Yeryomin wrote:
>>>> On 27 April 2011 20:19, Roman Yeryomin<leroi.lists at gmail.com>    
>>>> wrote:
>>>>> On 27 April 2011 17:26, Daniel-Constantin Mierla<miconda at gmail.com>
>>>>>   wrote:
>>>>>> Can you try with tcp disabled? i.e.,
>>>>>>
>>>>>> disable_tcp=yes
>>>>>>
>>>>>> in your config file.
>>>>>>
>>>>>> Might be the way the module is initialized for tcp handling.
>>>>>>
>>>>> it segfaults anyway - see new log in attachment
>>>>>
>>>> so.. any thoughts?
>>>>
>>> I am out of the office for a while, I had not time to look anymore 
>>> to it so
>>> far and probably unable to troubleshoot more during this time - I 
>>> have no
>>> testing environment setup for this module. Maybe someone else can 
>>> jump in,
>>> otherwise I will resume later.
>>>
>> Sorry for disturbing... is there any chance?
> it is, but I cannot promise a date yet -- too many things to catch up, 
> while testing this is very inconvenient not having a way to troubleshoot.
Not to forget about this one, is it possible to provide me access to a 
test machine where you have such configuration already set? Then I could 
just login and troubleshoot, far more faster than it will take to make 
myself such a testbed.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Fri Jun  3 09:19:48 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 09:19:48 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
In-Reply-To: <BANLkTinDze_Kk4rBbrtwEfy_Ub8WKWFDag@mail.gmail.com>
References: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
	<4DE64CBE.3030607@gmail.com>
	<BANLkTinDze_Kk4rBbrtwEfy_Ub8WKWFDag@mail.gmail.com>
Message-ID: <4DE88B14.7040603@gmail.com>

Hello,

did you get only one core file? This one seems from the main process 
when it discovered that a SIP worked crashed.

If you got more than one, send the backtraces from the other core files. 
If you got only one, please enable one core file per process and re-do 
it again.

See last part of:
http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles

Cheers,
Daniel


On 6/2/11 3:57 PM, Ozren Lapcevic wrote:
> Hi Daniel,
>
> here is the backtrace:
>
> #0  0x00007ffc46b81ed5 in raise () from /lib/libc.so.6
> #1  0x00007ffc46b833f3 in abort () from /lib/libc.so.6
> #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized 
> out>) at main.c:651
> #3 <signal handler called>
> #4  0x00007ffc46c1c307 in syscall () from /lib/libc.so.6
> #5  0x00007ffc4219f115 in imc_del_room (name=0x7ffc3fe0d388, 
> domain=0x7ffc3fe0d398) at ../../mem/../futexlock.h:110
> #6  0x00007ffc4219f1d8 in imc_htable_destroy () at imc_mng.c:104
> #7  0x00007ffc4219347a in destroy () at imc.c:736
> #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
> #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
> #10 0x0000000000463564 in shutdown_children (sig=<value optimized 
> out>, show_status=1) at main.c:668
> #11 0x0000000000463e9c in handle_sigs () at main.c:759
> #12 0x0000000000465572 in main_loop () at main.c:1676
> #13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) 
> at main.c:2398
>
> I'm not sure if I got the right thing. I got it with gdb 
> /usr/local/kamailio-ozren/sbin/kamailio --core core and called bt in gdb.
>
> I've repeated scenario described in the first mail: user tries to join 
> in a private room without getting an invite first.
>
> Best regards,
> Ozren
>
>
> On Wed, Jun 1, 2011 at 4:29 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     can you get the core dump and send the backtrace?
>
>     Be sure you have 'ulimit -c unlimited', working directory (-w
>     option) set to writable path and you don't run it in chowner (no
>     -u or -g parameters).
>
>     Cheers,
>     Daniel
>
>     On 6/1/11 4:25 PM, Ozren Lapcevic wrote:
>>     Hi,
>>
>>     there is a bug in IMC module that crashes Kamailio.
>>
>>     Here is the scenario: One user (pero at abc.hr <mailto:pero at abc.hr>)
>>     creates private chat room (chat-999 at abc.hr
>>     <mailto:chat-999 at abc.hr>). Other user (oz at abc.hr
>>     <mailto:oz at abc.hr>) tries to join the chat-999 at abc.hr
>>     <mailto:chat-999 at abc.hr> by sending #join. Kamailio crashes at
>>     that point. Here is the log:
>>
>>     Best regards,
>>     Ozren
>>
>>
>>     Jun  1 15:10:37 ser-test kernel: [5844312.456709]
>>     kamailio[20118]: segfault at 10 ip 7f28f89dae8e sp 7fffffffb750
>>     error 4 in imc.so[7f28f89cf000+17000]
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20118]: INFO: <script>:
>>     going to imc: F [sip:oz at abc.hr <mailto:sip%3Aoz at abc.hr>] T
>>     [sip:chat-999 at abc.hr <mailto:sip%3Achat-999 at abc.hr>] B [#join]
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core>
>>     [receive.c:186]: preparing to run routing scripts...
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc
>>     [imc.c:525]: found command
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: maxfwd
>>     [mf_funcs.c:85]: value = 70
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc
>>     [imc_cmd.c:146]: command: [join]
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core>
>>     [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=223
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc
>>     [imc_cmd.c:292]: found room [chat-999]
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core>
>>     [parser/msg_parser.c:103]: found end of header
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20118]: ERROR: imc
>>     [imc_cmd.c:317]: attept to join private room [chat-999] from user
>>     [oz]
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core>
>>     [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core>
>>     [main.c:741]: child process 20118 exited by a signal 11
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core>
>>     [parser/parse_to.c:803]: end of header reached, state=29
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core>
>>     [parser/msg_parser.c:630]: SIP Request:
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core>
>>     [main.c:744]: core was not generated
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core>
>>     [msg_translator.c:204]: check_via_address(161.53.255.82,
>>     161.53.255.82, 0)
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core>
>>     [parser/msg_parser.c:632]:  method: <MESSAGE>
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20113]: INFO: <core>
>>     [main.c:756]: INFO: terminating due to SIGCHLD
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20114]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20115]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20116]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20117]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20122]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20124]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20125]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20129]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20128]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20127]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20123]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20126]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20120]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20130]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20131]: INFO: <core>
>>     [main.c:807]: INFO: signal 15 received
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc
>>     [imc.c:709]: room 3 chat-777
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc
>>     [imc.c:709]: room 11 chat-888
>>     Jun  1 15:10:37 ser-test
>>     /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc
>>     [imc.c:709]: room 13 chat-999
>>
>>
>>
>>
>>
>>     _______________________________________________
>>     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://linkedin.com/in/miconda  -- http://twitter.com/miconda
>
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From miconda at gmail.com  Fri Jun  3 09:48:50 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 09:48:50 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
In-Reply-To: <4DE80BC1.3090903@sipwise.com>
References: <4DE7E343.1090509@sipwise.com> <4DE7F17C.7070709@gmail.com>
	<4DE7F743.4090501@sipwise.com> <4DE80A43.5000706@gmail.com>
	<4DE80BC1.3090903@sipwise.com>
Message-ID: <4DE891E2.1040802@gmail.com>

Hello,

I tested with kamailio default config and injected some REGISTERs to 
reproduce the scenario, see the SIP trace below. The results were ok in 
200 reply, also in kamctl.

Can you send me also a SIP trace for the situation? Also, the parameters 
for registrar and usrloc modules. Do you you have a particular 
matching_mode set?

Thanks,
Daniel

U 2011/06/03 09:39:05.409702 127.0.0.1:56741 -> 127.0.0.1:5060
REGISTER sip:127.0.0.1:5060 SIP/2.0
From: "Test" <sip:test at 127.0.0.1>;tag=as23
To: "Test" <sip:test at 127.0.0.1>
Call-ID: aa at 127.0.0.1
User-Agent: snom100_1.9b
CSeq: 2 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5080
Contact: <sip:me at 127.0.0.1:2001;ob>
Expires: 900
Content-Length: 0



U 2011/06/03 09:39:05.413813 127.0.0.1:5060 -> 127.0.0.1:5080
SIP/2.0 200 OK.
From: "Test" <sip:test at 127.0.0.1>;tag=as23
To: "Test" <sip:test at 127.0.0.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.521d
Call-ID: aa at 127.0.0.1
CSeq: 2 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5080.
Contact: <sip:me at 127.0.0.1:2001;ob>;expires=900.
Server: kamailio (3.2.0-dev5 (x86_64/darwin)).
Content-Length: 0.
.


U 2011/06/03 09:39:43.616036 127.0.0.1:56948 -> 127.0.0.1:5060
REGISTER sip:127.0.0.1:5060 SIP/2.0
From: "Test" <sip:test at 127.0.0.1>;tag=as23
To: "Test" <sip:test at 127.0.0.1>
Call-ID: aa at 127.0.0.1
User-Agent: snom100_1.9b
CSeq: 3 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5080
Contact: <sip:me at 127.0.0.1:3001;transport=UDP;ob>
Contact: <sip:me at 127.0.0.1:2001;ob>;expires=0
Expires: 900
Content-Length: 0



U 2011/06/03 09:39:43.618327 127.0.0.1:5060 -> 127.0.0.1:5080
SIP/2.0 200 OK.
From: "Test" <sip:test at 127.0.0.1>;tag=as23
To: "Test" <sip:test at 127.0.0.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.521d
Call-ID: aa at 127.0.0.1
CSeq: 3 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5080.
Contact: <sip:me at 127.0.0.1:3001;transport=UDP;ob>;expires=900.
Server: kamailio (3.2.0-dev5 (x86_64/darwin)).
Content-Length: 0.
.

# kamctl ul show
Domain:: location table=512 records=1 max_slot=1
     AOR:: test
         Contact:: sip:me at 127.0.0.1:3001;transport=UDP;ob Q=
             Expires:: 889
             Callid:: aa at 127.0.0.1
             Cseq:: 3
             User-agent:: snom100_1.9b
             State:: CS_NEW
             Flags:: 0
             Cflag:: 0
             Socket:: udp:127.0.0.1:5060
             Methods:: 4294967295

# kamctl ul show
Domain:: location table=512 records=1 max_slot=1
     AOR:: test
         Contact:: sip:me at 127.0.0.1:3001;transport=UDP;ob Q=
             Expires:: 690
             Callid:: aa at 127.0.0.1
             Cseq:: 3
             User-agent:: snom100_1.9b
             State:: CS_NEW
             Flags:: 0
             Cflag:: 0
             Socket:: udp:127.0.0.1:5060
             Methods:: 4294967295

On 6/3/11 12:16 AM, Andreas Granig wrote:
> Hi,
>
> On 06/03/2011 12:10 AM, Daniel-Constantin Mierla wrote:
>> one more thing, what is the output of kamctl ul show? I want to see if
>> it is a matter of building the contact header for reply or is in
>> internal list of location records.
> Domain:: location table=512 records=1 max_slot=1
> 	AOR:: me at somedomain.com
> 		Contact:: sip:me at 10.0.0.7:52030;ob Q=
> 			Expires:: 875
> 			Callid:: DZZHTDttFd5WivydjNnWfpRLsXp8RIYh
> 			Cseq:: 4599
> 			User-agent:: myclient
> 			Received:: sip:1.2.3.4:55642
> 			Path::<sip:lb at 127.0.0.1:5060;lr;received=sip:1.2.3.4:55642>
> 			State:: CS_SYNC
> 			Flags:: 0
> 			Cflag:: 192
> 			Socket:: udp:127.0.0.1:5062
> 			Methods:: 8143
> 		Contact:: sip:me at 1.2.3.4:55642;transport=UDP;ob Q=
> 			Expires:: 876
> 			Callid:: DZZHTDttFd5WivydjNnWfpRLsXp8RIYh
> 			Cseq:: 4601
> 			User-agent:: myclient
> 			Path::<sip:lb at 127.0.0.1:5060;lr>
> 			State:: CS_SYNC
> 			Flags:: 0
> 			Cflag:: 0
> 			Socket:: udp:127.0.0.1:5062
> 			Methods:: 8143
>
> Hope this helps,
> Andreas

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ozren.lapcevic at gmail.com  Fri Jun  3 09:59:31 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Fri, 3 Jun 2011 09:59:31 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
In-Reply-To: <4DE88B14.7040603@gmail.com>
References: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
	<4DE64CBE.3030607@gmail.com>
	<BANLkTinDze_Kk4rBbrtwEfy_Ub8WKWFDag@mail.gmail.com>
	<4DE88B14.7040603@gmail.com>
Message-ID: <BANLkTi=0qzp-qXAQQMS4WJ8pHtu_vLDpVA@mail.gmail.com>

Hi,

I previously got only one core file, but now that I've enabled one core file
per process, I got two core files. Here are the backtraces:

#0  0x00007fc9c5787ed5 in raise () from /lib/libc.so.6
#1  0x00007fc9c57893f3 in abort () from /lib/libc.so.6
#2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
main.c:651
#3  <signal handler called>
#4  0x00007fc9c5822307 in syscall () from /lib/libc.so.6
#5  0x00007fc9c0da50c6 in imc_del_room (name=0x7fc9bea13438,
domain=0x7fc9bea13448) at ../../mem/../futexlock.h:123
#6  0x00007fc9c0da51d8 in imc_htable_destroy () at imc_mng.c:104
#7  0x00007fc9c0d9947a in destroy () at imc.c:736
#8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
#9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
#10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
show_status=1) at main.c:668
#11 0x0000000000463e9c in handle_sigs () at main.c:759
#12 0x0000000000465572 in main_loop () at main.c:1676
#13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
main.c:2398

and:

#0  imc_handle_join (msg=<value optimized out>, cmd=<value optimized out>,
src=<value optimized out>, dst=<value optimized out>) at imc_cmd.c:329
#1  0x00007fc9c0d985b0 in imc_manager (msg=0x91ce20, str1=<value optimized
out>, str2=<value optimized out>) at imc.c:544
#2  0x000000000041591f in do_action (h=0x7fffffffad30, a=0x925c28,
msg=0x91ce20) at action.c:1095
#3  0x000000000041cc5e in run_actions (h=0x7fffffffad30, a=0x925c28,
msg=0x91ce20) at action.c:1555
#4  0x000000000041cf97 in run_actions_safe (h=0x7fffffffbdb0,
a=0x7fc9c5873798, msg=0x0) at action.c:1607
#5  0x00000000004b1574 in rval_get_int (h=0x7fffffffbdb0, msg=<value
optimized out>, i=0x7fffffffb408, rv=0xd1d990, cache=0xd1d910) at
rvalue.c:904
#6  0x00000000004b4fdc in rval_expr_eval_int (h=0x7fffffffbdb0,
msg=0x91ce20, res=0x7fffffffb408, rve=0x9262c0) at rvalue.c:1866
#7  0x0000000000416e02 in do_action (h=0x7fffffffbdb0, a=0x927000,
msg=0x91ce20) at action.c:1071
#8  0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x9258a0,
msg=0x91ce20) at action.c:1555
#9  0x0000000000416e60 in do_action (h=0x7fffffffbdb0, a=0x927ab8,
msg=0x91ce20) at action.c:1090
#10 0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x921c38,
msg=0x91ce20) at action.c:1555
#11 0x00000000004177e5 in do_action (h=0x7fffffffbdb0, a=0x90cc40,
msg=0x91ce20) at action.c:711
#12 0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x9079e0,
msg=0x91ce20) at action.c:1555
#13 0x000000000041cf23 in run_top_route (a=0x9079e0, msg=0x91ce20, c=<value
optimized out>) at action.c:1628
#14 0x00000000004958c6 in receive_msg (
    buf=0x8a3a00 "MESSAGE sip:chat-999 at abc.hr SIP/2.0\r\nCall-ID:
9d5779ef64ae7394e7b922cc4e98e424 at 0.0.0.0\r\nCSeq: 11051797 MESSAGE\r\nFrom:
\"oz\" <sip:oz at abc.hr>;tag=6d7f9acd\r\nTo: <sip:chat-999 at abc.hr>\r\nVia:
SIP/2.0/UDP 161"..., len=<value optimized out>, rcv_info=0x7fffffffc030) at
receive.c:205
#15 0x000000000051c861 in udp_rcv_loop () at udp_server.c:532
#16 0x0000000000464b15 in main_loop () at main.c:1554
#17 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
main.c:2398


Best regards,
Ozren


On Fri, Jun 3, 2011 at 9:19 AM, Daniel-Constantin Mierla
<miconda at gmail.com>wrote:

>  Hello,
>
> did you get only one core file? This one seems from the main process when
> it discovered that a SIP worked crashed.
>
> If you got more than one, send the backtraces from the other core files. If
> you got only one, please enable one core file per process and re-do it
> again.
>
> See last part of:
> http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles
>
> Cheers,
> Daniel
>
>
>
> On 6/2/11 3:57 PM, Ozren Lapcevic wrote:
>
> Hi Daniel,
>
> here is the backtrace:
>
> #0  0x00007ffc46b81ed5 in raise () from /lib/libc.so.6
> #1  0x00007ffc46b833f3 in abort () from /lib/libc.so.6
> #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
> main.c:651
> #3  <signal handler called>
> #4  0x00007ffc46c1c307 in syscall () from /lib/libc.so.6
> #5  0x00007ffc4219f115 in imc_del_room (name=0x7ffc3fe0d388,
> domain=0x7ffc3fe0d398) at ../../mem/../futexlock.h:110
> #6  0x00007ffc4219f1d8 in imc_htable_destroy () at imc_mng.c:104
> #7  0x00007ffc4219347a in destroy () at imc.c:736
> #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
> #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
> #10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
> show_status=1) at main.c:668
> #11 0x0000000000463e9c in handle_sigs () at main.c:759
> #12 0x0000000000465572 in main_loop () at main.c:1676
> #13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
> main.c:2398
>
> I'm not sure if I got the right thing. I got it with gdb
> /usr/local/kamailio-ozren/sbin/kamailio --core core and called bt in gdb.
>
> I've repeated scenario described in the first mail: user tries to join in a
> private room without getting an invite first.
>
> Best regards,
> Ozren
>
>
> On Wed, Jun 1, 2011 at 4:29 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> can you get the core dump and send the backtrace?
>>
>> Be sure you have 'ulimit -c unlimited', working directory (-w option) set
>> to writable path and you don't run it in chowner (no -u or -g parameters).
>>
>> Cheers,
>> Daniel
>>
>> On 6/1/11 4:25 PM, Ozren Lapcevic wrote:
>>
>>  Hi,
>>
>> there is a bug in IMC module that crashes Kamailio.
>>
>> Here is the scenario: One user (pero at abc.hr) creates private chat room (
>> chat-999 at abc.hr). Other user (oz at abc.hr) tries to join the
>> chat-999 at abc.hr by sending #join. Kamailio crashes at that point. Here is
>> the log:
>>
>> Best regards,
>> Ozren
>>
>>
>> Jun  1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]:
>> segfault at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in
>> imc.so[7f28f89cf000+17000]
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-999 at abc.hr] B
>> [#join]
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: <core> [receive.c:186]: preparing to run routing scripts...
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>> DEBUG: imc [imc.c:525]: found command
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: maxfwd [mf_funcs.c:85]: value = 70
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>> DEBUG: imc [imc_cmd.c:146]: command: [join]
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body :
>> content_length=223
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>> DEBUG: imc [imc_cmd.c:292]: found room [chat-999]
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: <core> [parser/msg_parser.c:103]: found end of header
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>> ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user
>> [oz]
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>> ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
>> DEBUG: <core> [parser/msg_parser.c:630]: SIP Request:
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>> ALERT: <core> [main.c:744]: core was not generated
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>> DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82,
>> 161.53.255.82, 0)
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
>> DEBUG: <core> [parser/msg_parser.c:632]:  method:  <MESSAGE>
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>> INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]:
>> INFO: <core> [main.c:807]: INFO: signal 15 received
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>> DEBUG: imc [imc.c:709]: room 3 chat-777
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>> DEBUG: imc [imc.c:709]: room 11 chat-888
>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>> DEBUG: imc [imc.c:709]: room 13 chat-999
>>
>>
>>
>>
>>
>>  _______________________________________________
>> 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://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From ozren.lapcevic at gmail.com  Fri Jun  3 10:19:45 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Fri, 3 Jun 2011 10:19:45 +0200
Subject: [SR-Users] IMC module unresolved issues
In-Reply-To: <BANLkTimUq4kJqqRsDcLQT733xRDdMasNoA@mail.gmail.com>
References: <BANLkTimt1T2f08+Qu2mO4iPDXKqdtq48oA@mail.gmail.com>
	<4DE6173D.6000409@gmail.com>
	<BANLkTikmtRezEkXE+wAz1oYxSV=FXGR65w@mail.gmail.com>
	<4DE64A4C.7090201@gmail.com>
	<BANLkTimUq4kJqqRsDcLQT733xRDdMasNoA@mail.gmail.com>
Message-ID: <BANLkTinM=gvCHO_VQcxpFbc=7mP_yYvvAg@mail.gmail.com>

Hi,

I'm resending issue #1 described in previous mail, with backtraces from both
the main process and the child process (previously I had posted only the
main process) and log which is slightly different.

#Call of ./kamctl fifo imc_list_rooms
Jun  3 10:09:13 ser-test ./kamailio[15794]: : <core> [pass_fd.c:293]: ERROR:
receive_fd: EOF on 23
Jun  3 10:09:13 ser-test ./kamailio[15794]: DEBUG: <core> [tcp_main.c:3554]:
DBG: handle_ser_child: dead child 11, pid 15787 (shutting down?)
Jun  3 10:09:13 ser-test ./kamailio[15794]: DEBUG: <core> [io_wait.h:617]:
DBG: io_watch_del (0x81b400, 23, -1, 0x0) fd_no=23 called
Jun  3 10:09:13 ser-test ./kamailio[15776]: ALERT: <core> [main.c:741]:
child process 15787 exited by a signal 11
Jun  3 10:09:13 ser-test ./kamailio[15776]: ALERT: <core> [main.c:744]: core
was generated
Jun  3 10:09:13 ser-test ./kamailio[15776]: INFO: <core> [main.c:756]: INFO:
terminating due to SIGCHLD
Jun  3 10:09:13 ser-test ./kamailio[15777]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15778]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15779]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15780]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15781]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15782]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15783]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15784]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15785]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15788]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15789]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15790]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15791]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15786]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15792]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15793]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15794]: INFO: <core> [main.c:807]: INFO:
signal 15 received
Jun  3 10:09:13 ser-test ./kamailio[15776]: DEBUG: imc [imc.c:709]: room 11
chat-555
Jun  3 10:10:13 ser-test ./kamailio[15776]: : <core> [main.c:650]: BUG:
shutdown timeout triggered, dying...

Backtraces:
#0  0x00007f6441657ed5 in raise () from /lib/libc.so.6
#1  0x00007f64416593f3 in abort () from /lib/libc.so.6
#2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
main.c:651
#3  <signal handler called>
#4  0x00007f64416f2307 in syscall () from /lib/libc.so.6
#5  0x00007f643cc750c6 in imc_del_room (name=0x7f643a902100,
domain=0x7f643a902110) at ../../mem/../futexlock.h:123
#6  0x00007f643cc751d8 in imc_htable_destroy () at imc_mng.c:104
#7  0x00007f643cc6947a in destroy () at imc.c:736
#8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
#9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
#10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
show_status=1) at main.c:668
#11 0x0000000000463e9c in handle_sigs () at main.c:759
#12 0x0000000000465572 in main_loop () at main.c:1676
#13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
main.c:2398

#0  0x00007f643cc6ba46 in imc_mi_list_rooms (cmd_tree=<value optimized out>,
param=<value optimized out>) at imc.c:775
#1  0x00007f643febe694 in mi_fifo_server (fifo_stream=0xce66d0) at
../../lib/kmi/mi.h:77
#2  0x00007f643fec08bb in fifo_process (rank=<value optimized out>) at
mi_fifo.c:244
#3  0x00007f643fec0bda in mi_child_init (rank=0) at mi_fifo.c:208
#4  0x00000000004d8c43 in init_mod_child (m=0x8f4d08, rank=0) at
sr_module.c:829
#5  0x00000000004d8c1c in init_mod_child (m=0x8f4f78, rank=0) at
sr_module.c:826
#6  0x00000000004d8c1c in init_mod_child (m=0x8f5600, rank=0) at
sr_module.c:826
#7  0x00000000004d8c1c in init_mod_child (m=0x8f6da8, rank=0) at
sr_module.c:826
#8  0x00000000004d8c1c in init_mod_child (m=0x8f74f0, rank=0) at
sr_module.c:826
#9  0x00000000004d8c1c in init_mod_child (m=0x8f79f8, rank=0) at
sr_module.c:826
#10 0x00000000004d8c1c in init_mod_child (m=0x8f8168, rank=0) at
sr_module.c:826
#11 0x00000000004d8c1c in init_mod_child (m=0x8fd1c8, rank=0) at
sr_module.c:826
#12 0x00000000004d8c1c in init_mod_child (m=0x8fd540, rank=0) at
sr_module.c:826
#13 0x00000000004d8c1c in init_mod_child (m=0x8fd888, rank=0) at
sr_module.c:826
#14 0x00000000004d8c1c in init_mod_child (m=0x8fde90, rank=0) at
sr_module.c:826
#15 0x00000000004d8c1c in init_mod_child (m=0x8fe908, rank=0) at
sr_module.c:826
#16 0x00000000004d8c1c in init_mod_child (m=0x8ff038, rank=0) at
sr_module.c:826
#17 0x00000000004d8c1c in init_mod_child (m=0x8ff5f0, rank=0) at
sr_module.c:826
#18 0x00000000004d8c1c in init_mod_child (m=0x8ff900, rank=0) at
sr_module.c:826
#19 0x00000000004d8c1c in init_mod_child (m=0x8fff70, rank=0) at
sr_module.c:826
#20 0x00000000004d8c1c in init_mod_child (m=0x9004b8, rank=0) at
sr_module.c:826
#21 0x00000000004d8c1c in init_mod_child (m=0x900910, rank=0) at
sr_module.c:826
#22 0x00000000004d8c1c in init_mod_child (m=0x900d38, rank=0) at
sr_module.c:826
#23 0x00000000004d8c1c in init_mod_child (m=0x901120, rank=0) at
sr_module.c:826
#24 0x00000000004d8c1c in init_mod_child (m=0x901510, rank=0) at
sr_module.c:826
#25 0x00000000004d8c1c in init_mod_child (m=0x901b80, rank=0) at
sr_module.c:826
#26 0x00000000004d8c1c in init_mod_child (m=0x9022c0, rank=0) at
sr_module.c:826
#27 0x00000000004654d5 in main_loop () at main.c:1624
#28 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
main.c:2398

Best regards,
Ozren


On Thu, Jun 2, 2011 at 4:54 PM, Ozren Lapcevic <ozren.lapcevic at gmail.com>wrote:

> On Wed, Jun 1, 2011 at 4:18 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>
>> Any issues still going on with the module?
>>
>> Best regards,
>> Daniel
>>
>
>
> Hi Daniel,
>
> yes, there are still several issues we've found. I hope that you or the
> other authors of the module will find some time to look into the issues.
>
> 1. last member in the chat room exits + call of ./kamctl fifo
> imc_list_rooms crashes Kamailio.
> Working example (no bugs): oz at abc.hr creates chat room chat-555 at abc.hr.
> Doesn't write messages to the chat room. Exits (#exit). Receives message:
> "The room has been destroyed". Call of ./kamctl fifo imc_list_rooms works
> fine and doesn't display the room which was just destroyed.
>
> Example which crashes Kamailio: oz at abc.hr creates chat room
> chat-555 at abc.hr. Writes something to the chat room. Exits (#exit). No
> messages are displayed in SIP client, although he should get "The room has
> been destroyed". Call of ./kamctl fifo imc_list_rooms doesn't display
> anything (although there are other rooms besides chat-555 at abc.hr) and
> crashes Kamailio. In the database, in imc_rooms, chat-555 at abc.hr is still
> there, but there are no members in imc_members table. Log:
> #processing of #exit
>
> Jun  2 16:21:49 ser-test ./kamailio[5547]: INFO: <script>: going to imc: F
> [sip:oz at abc.hr] T [sip:chat-555 at abc.hr] B [#exit]
> Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc.c:525]: found
> command
> Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc_cmd.c:146]:
> command: [exit]
> Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc_mng.c:373]:
> found member
> Jun  2 16:21:49 ser-test ./kamailio[5547]: DEBUG: imc [imc_cmd.c:1091]: nr
> = 0
>
> #Call of ./kamctl fifo imc_list_rooms
>
> Jun  2 16:22:02 ser-test ./kamailio[5550]: DEBUG: mi_fifo [fifo_fnc.c:493]:
> entered consume
> Jun  2 16:22:02 ser-test kernel: [5934996.535094] kamailio[5550]: segfault
> at 10 ip 7f85c0b92a46 sp 7fffffffc700 error 4 in imc.so[7f85c0b8c000+17000]
> Jun  2 16:22:02 ser-test ./kamailio[5550]: DEBUG: mi_fifo [fifo_fnc.c:493]:
> **** done consume
> Jun  2 16:22:02 ser-test ./kamailio[5550]: DEBUG: mi_fifo [fifo_fnc.c:507]:
> done parsing the mi tree
> Jun  2 16:22:02 ser-test ./kamailio[5557]: : <core> [pass_fd.c:293]: ERROR:
> receive_fd: EOF on 23
> Jun  2 16:22:02 ser-test ./kamailio[5557]: DEBUG: <core> [tcp_main.c:3554]:
> DBG: handle_ser_child: dead child 11, pid 5550 (shutting down?)
> Jun  2 16:22:02 ser-test ./kamailio[5557]: DEBUG: <core> [io_wait.h:617]:
> DBG: io_watch_del (0x81b400, 23, -1, 0x0) fd_no=23 called
> Jun  2 16:22:02 ser-test ./kamailio[5539]: ALERT: <core> [main.c:741]:
> child process 5550 exited by a signal 11
> Jun  2 16:22:02 ser-test ./kamailio[5539]: ALERT: <core> [main.c:744]: core
> was generated
> Jun  2 16:22:02 ser-test ./kamailio[5539]: INFO: <core> [main.c:756]: INFO:
> terminating due to SIGCHLD
> Jun  2 16:22:02 ser-test ./kamailio[5540]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5541]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5542]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5543]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5544]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5545]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5546]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5547]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5548]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5551]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5552]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5553]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5549]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5554]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5555]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5556]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5557]: INFO: <core> [main.c:807]: INFO:
> signal 15 received
> Jun  2 16:22:02 ser-test ./kamailio[5539]: DEBUG: imc [imc.c:709]: room 11
> chat-555
>
> Backtrace: (same as for #join to private conference described in other
> topic!)
> #0  0x00007f85c557eed5 in raise () from /lib/libc.so.6
> #1  0x00007f85c55803f3 in abort () from /lib/libc.so.6
> #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
> main.c:651
> #3  <signal handler called>
> #4  0x00007f85c5619307 in syscall () from /lib/libc.so.6
> #5  0x00007f85c0b9c0c6 in imc_del_room (name=0x7f85be80a388,
> domain=0x7f85be80a398) at ../../mem/../futexlock.h:123
> #6  0x00007f85c0b9c1d8 in imc_htable_destroy () at imc_mng.c:104
> #7  0x00007f85c0b9047a in destroy () at imc.c:736
> #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
> #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
> #10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
> show_status=1) at main.c:668
> #11 0x0000000000463e9c in handle_sigs () at main.c:759
> #12 0x0000000000465572 in main_loop () at main.c:1676
> #13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
> main.c:2398
>
>
> 2. #remove command prints wrong info.
> Example: oz at abc.hr creates chat room chat-555 at abc.hr. Invites
> slavko at abc.hr (#invite slavko at abc.hr). Invites pero at abc.hr. Both of them
> join with #accept. oz at abc.hr removes pero (#remove pero at abc.hr). Pero's
> client displays messages: "You have been removed from this room." and Pero
> can't send messages to chat room any more. However, Oz's and Slavko's
> clients display message: "<sip:pero at abc.hr> has joined the room".
>
>
> 3. #invite & #remove commands do not work after typing some text in
> conference room.
> Example 1: oz at abc.hr creates chat room chat-555 at abc.hr. Invites
> slavko at abc.hr (#invite slavko at abc.hr). Invites pero at abc.hr. Both of them
> join with #accept. Pero and Slavko write messages to chat room. Oz can
> remove them at this point. However, after Oz writes a message to chat room,
> he can't remove members any more and receives 500 command error. Log:
>
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-555 at abc.hr] B
> [#remove slavko at abc.hr]
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> DEBUG: imc [imc.c:525]: found command
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> DEBUG: imc [imc_cmd.c:146]: command: [remove]
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> DEBUG: imc [imc_cmd.c:151]: parameter 0=[slavko at abc.hr]
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> DEBUG: imc [imc_mng.c:373]: found member
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> ERROR: imc [imc_cmd.c:664]: user [oz] has no right to remove other users [
> sip:slavko at abc.hr]!
> Jun  2 14:58:42 ser-test /usr/local/kamailio-ozren/sbin/kamailio[4139]:
> ERROR: imc [imc.c:578]: failed to handle 'remove'
>
>
> ./kamctl fifo imc_list_rooms
> ROOM::  URI=sip:chat-555 at abc.hr MEMBERS=0 OWNER=sip:oz at abc.hr
>
> ./kamctl fifo imc_list_members sip:chat-555 at abc.hr
> ROOM:: sip:chat-555 at abc.hr NR_OF_MEMBERS=2
>
>         MEMBER:: sip:oz at abc.hr
>         MEMBER:: sip:slavko at abc.hr
>
> Example 2: oz at abc.hr creates chat room chat-666 at abc.hr. Invites
> slavko at abc.hr. Slavko joins with #accept. Oz writes a message to chat
> room. Invites pero at abc.hr. Imc manager returns 500 command error. (NOTE:
> If Oz doesn't write any text to chat room, he can invite multiple users.)
> Here is the log:
>
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-666 at abc.hr] B
> [#invite pero at abc.hr]
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> DEBUG: imc [imc.c:525]: found command
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> DEBUG: imc [imc_cmd.c:146]: command: [invite]
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> DEBUG: imc [imc_cmd.c:151]: parameter 0=[pero at abc.hr]
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> DEBUG: imc [imc_mng.c:373]: found member
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> ERROR: imc [imc_cmd.c:441]: user [oz] has no right to invite other users!
> Jun  2 16:42:43 ser-test /usr/local/kamailio-ozren/sbin/kamailio[5848]:
> ERROR: imc [imc.c:554]: failed to handle 'invite'
>
>
> ./kamctl fifo imc_list_rooms
> ROOM::  URI=sip:chat-666 at abc.hr MEMBERS=0 OWNER=sip:oz at abc.hr
>
> ./kamctl fifo imc_list_members sip:chat-666 at abc.hr
> ROOM:: sip:chat-666 at abc.hr NR_OF_MEMBERS=2
>
>         MEMBER:: sip:oz at abc.hr
>         MEMBER:: sip:slavko at abc.hr
>
>
> 4. ./kamctl fifo imc_list_rooms command doesn't report correct number of
> members in the room, number of members remains 0, although members have
> successfully joined the chat room and ./kamctl fifo imc_list_members
> sip:chat-555 at abc.hr returns correct number of members.
>
>
> Best regards,
> Ozren
>
>
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From timo.reimann at 1und1.de  Fri Jun  3 10:39:09 2011
From: timo.reimann at 1und1.de (Timo Reimann)
Date: Fri, 3 Jun 2011 10:39:09 +0200
Subject: [SR-Users] callback function of tm module
In-Reply-To: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>
References: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>
Message-ID: <4DE89DAD.3060501@1und1.de>

Hi Bruno,


On 03.06.2011 00:07, Bruno Bresciani wrote:
> I'm having doubts in the implementation of the callback function module
> tm. As she calls the function to record register_tmcb ()?
> Can someome help me?

Could you be more specific on which part of the tm module you are having
trouble with?

If you need an example on how to use tm's callbacks you may take a look
at the dialog module, specifically the files dialog.c and dlg_handlers.c.


Cheers,

--Timo


From miconda at gmail.com  Fri Jun  3 10:42:39 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 10:42:39 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
In-Reply-To: <BANLkTi=0qzp-qXAQQMS4WJ8pHtu_vLDpVA@mail.gmail.com>
References: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
	<4DE64CBE.3030607@gmail.com>
	<BANLkTinDze_Kk4rBbrtwEfy_Ub8WKWFDag@mail.gmail.com>
	<4DE88B14.7040603@gmail.com>
	<BANLkTi=0qzp-qXAQQMS4WJ8pHtu_vLDpVA@mail.gmail.com>
Message-ID: <4DE89E7F.3080008@gmail.com>

Hello,

I did a fix for the case of joining private rooms without permissions, 
can you try it again (pull the git branch 3.1)?

Thanks,
Daniel


On 6/3/11 9:59 AM, Ozren Lapcevic wrote:
> Hi,
>
> I previously got only one core file, but now that I've enabled one 
> core file per process, I got two core files. Here are the backtraces:
>
> #0  0x00007fc9c5787ed5 in raise () from /lib/libc.so.6
> #1  0x00007fc9c57893f3 in abort () from /lib/libc.so.6
> #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized 
> out>) at main.c:651
> #3 <signal handler called>
> #4  0x00007fc9c5822307 in syscall () from /lib/libc.so.6
> #5  0x00007fc9c0da50c6 in imc_del_room (name=0x7fc9bea13438, 
> domain=0x7fc9bea13448) at ../../mem/../futexlock.h:123
> #6  0x00007fc9c0da51d8 in imc_htable_destroy () at imc_mng.c:104
> #7  0x00007fc9c0d9947a in destroy () at imc.c:736
> #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
> #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
> #10 0x0000000000463564 in shutdown_children (sig=<value optimized 
> out>, show_status=1) at main.c:668
> #11 0x0000000000463e9c in handle_sigs () at main.c:759
> #12 0x0000000000465572 in main_loop () at main.c:1676
> #13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) 
> at main.c:2398
>
> and:
>
> #0  imc_handle_join (msg=<value optimized out>, cmd=<value optimized 
> out>, src=<value optimized out>, dst=<value optimized out>) at 
> imc_cmd.c:329
> #1  0x00007fc9c0d985b0 in imc_manager (msg=0x91ce20, str1=<value 
> optimized out>, str2=<value optimized out>) at imc.c:544
> #2  0x000000000041591f in do_action (h=0x7fffffffad30, a=0x925c28, 
> msg=0x91ce20) at action.c:1095
> #3  0x000000000041cc5e in run_actions (h=0x7fffffffad30, a=0x925c28, 
> msg=0x91ce20) at action.c:1555
> #4  0x000000000041cf97 in run_actions_safe (h=0x7fffffffbdb0, 
> a=0x7fc9c5873798, msg=0x0) at action.c:1607
> #5  0x00000000004b1574 in rval_get_int (h=0x7fffffffbdb0, msg=<value 
> optimized out>, i=0x7fffffffb408, rv=0xd1d990, cache=0xd1d910) at 
> rvalue.c:904
> #6  0x00000000004b4fdc in rval_expr_eval_int (h=0x7fffffffbdb0, 
> msg=0x91ce20, res=0x7fffffffb408, rve=0x9262c0) at rvalue.c:1866
> #7  0x0000000000416e02 in do_action (h=0x7fffffffbdb0, a=0x927000, 
> msg=0x91ce20) at action.c:1071
> #8  0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x9258a0, 
> msg=0x91ce20) at action.c:1555
> #9  0x0000000000416e60 in do_action (h=0x7fffffffbdb0, a=0x927ab8, 
> msg=0x91ce20) at action.c:1090
> #10 0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x921c38, 
> msg=0x91ce20) at action.c:1555
> #11 0x00000000004177e5 in do_action (h=0x7fffffffbdb0, a=0x90cc40, 
> msg=0x91ce20) at action.c:711
> #12 0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x9079e0, 
> msg=0x91ce20) at action.c:1555
> #13 0x000000000041cf23 in run_top_route (a=0x9079e0, msg=0x91ce20, 
> c=<value optimized out>) at action.c:1628
> #14 0x00000000004958c6 in receive_msg (
>     buf=0x8a3a00 "MESSAGE sip:chat-999 at abc.hr 
> <mailto:sip%3Achat-999 at abc.hr> SIP/2.0\r\nCall-ID: 
> 9d5779ef64ae7394e7b922cc4e98e424 at 0.0.0.0 
> <mailto:9d5779ef64ae7394e7b922cc4e98e424 at 0.0.0.0>\r\nCSeq: 11051797 
> MESSAGE\r\nFrom: \"oz\" <sip:oz at abc.hr 
> <mailto:sip%3Aoz at abc.hr>>;tag=6d7f9acd\r\nTo: <sip:chat-999 at abc.hr 
> <mailto:sip%3Achat-999 at abc.hr>>\r\nVia: SIP/2.0/UDP 161"..., 
> len=<value optimized out>, rcv_info=0x7fffffffc030) at receive.c:205
> #15 0x000000000051c861 in udp_rcv_loop () at udp_server.c:532
> #16 0x0000000000464b15 in main_loop () at main.c:1554
> #17 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) 
> at main.c:2398
>
>
> Best regards,
> Ozren
>
>
> On Fri, Jun 3, 2011 at 9:19 AM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     did you get only one core file? This one seems from the main
>     process when it discovered that a SIP worked crashed.
>
>     If you got more than one, send the backtraces from the other core
>     files. If you got only one, please enable one core file per
>     process and re-do it again.
>
>     See last part of:
>     http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles
>
>     Cheers,
>     Daniel
>
>
>
>     On 6/2/11 3:57 PM, Ozren Lapcevic wrote:
>>     Hi Daniel,
>>
>>     here is the backtrace:
>>
>>     #0  0x00007ffc46b81ed5 in raise () from /lib/libc.so.6
>>     #1  0x00007ffc46b833f3 in abort () from /lib/libc.so.6
>>     #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized
>>     out>) at main.c:651
>>     #3 <signal handler called>
>>     #4  0x00007ffc46c1c307 in syscall () from /lib/libc.so.6
>>     #5  0x00007ffc4219f115 in imc_del_room (name=0x7ffc3fe0d388,
>>     domain=0x7ffc3fe0d398) at ../../mem/../futexlock.h:110
>>     #6  0x00007ffc4219f1d8 in imc_htable_destroy () at imc_mng.c:104
>>     #7  0x00007ffc4219347a in destroy () at imc.c:736
>>     #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
>>     #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
>>     #10 0x0000000000463564 in shutdown_children (sig=<value optimized
>>     out>, show_status=1) at main.c:668
>>     #11 0x0000000000463e9c in handle_sigs () at main.c:759
>>     #12 0x0000000000465572 in main_loop () at main.c:1676
>>     #13 0x0000000000467f53 in main (argc=<value optimized out>,
>>     argv=0x0) at main.c:2398
>>
>>     I'm not sure if I got the right thing. I got it with gdb
>>     /usr/local/kamailio-ozren/sbin/kamailio --core core and called bt
>>     in gdb.
>>
>>     I've repeated scenario described in the first mail: user tries to
>>     join in a private room without getting an invite first.
>>
>>     Best regards,
>>     Ozren
>>
>>
>>     On Wed, Jun 1, 2011 at 4:29 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         can you get the core dump and send the backtrace?
>>
>>         Be sure you have 'ulimit -c unlimited', working directory (-w
>>         option) set to writable path and you don't run it in chowner
>>         (no -u or -g parameters).
>>
>>         Cheers,
>>         Daniel
>>
>>         On 6/1/11 4:25 PM, Ozren Lapcevic wrote:
>>>         Hi,
>>>
>>>         there is a bug in IMC module that crashes Kamailio.
>>>
>>>         Here is the scenario: One user (pero at abc.hr
>>>         <mailto:pero at abc.hr>) creates private chat room
>>>         (chat-999 at abc.hr <mailto:chat-999 at abc.hr>). Other user
>>>         (oz at abc.hr <mailto:oz at abc.hr>) tries to join the
>>>         chat-999 at abc.hr <mailto:chat-999 at abc.hr> by sending #join.
>>>         Kamailio crashes at that point. Here is the log:
>>>
>>>         Best regards,
>>>         Ozren
>>>
>>>
>>>         Jun  1 15:10:37 ser-test kernel: [5844312.456709]
>>>         kamailio[20118]: segfault at 10 ip 7f28f89dae8e sp
>>>         7fffffffb750 error 4 in imc.so[7f28f89cf000+17000]
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20118]: INFO:
>>>         <script>: going to imc: F [sip:oz at abc.hr
>>>         <mailto:sip%3Aoz at abc.hr>] T [sip:chat-999 at abc.hr
>>>         <mailto:sip%3Achat-999 at abc.hr>] B [#join]
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         <core> [receive.c:186]: preparing to run routing scripts...
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc
>>>         [imc.c:525]: found command
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         maxfwd [mf_funcs.c:85]: value = 70
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc
>>>         [imc_cmd.c:146]: command: [join]
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body :
>>>         content_length=223
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc
>>>         [imc_cmd.c:292]: found room [chat-999]
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         <core> [parser/msg_parser.c:103]: found end of header
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20118]: ERROR: imc
>>>         [imc_cmd.c:317]: attept to join private room [chat-999] from
>>>         user [oz]
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT:
>>>         <core> [main.c:741]: child process 20118 exited by a signal 11
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         <core> [parser/parse_to.c:803]: end of header reached, state=29
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG:
>>>         <core> [parser/msg_parser.c:630]: SIP Request:
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT:
>>>         <core> [main.c:744]: core was not generated
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG:
>>>         <core> [msg_translator.c:204]:
>>>         check_via_address(161.53.255.82, 161.53.255.82, 0)
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG:
>>>         <core> [parser/msg_parser.c:632]:  method: <MESSAGE>
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20113]: INFO: <core>
>>>         [main.c:756]: INFO: terminating due to SIGCHLD
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20114]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20115]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20116]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20117]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20122]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20124]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20125]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20129]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20128]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20127]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20123]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20126]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20120]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20130]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20131]: INFO: <core>
>>>         [main.c:807]: INFO: signal 15 received
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc
>>>         [imc.c:709]: room 3 chat-777
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc
>>>         [imc.c:709]: room 11 chat-888
>>>         Jun  1 15:10:37 ser-test
>>>         /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc
>>>         [imc.c:709]: room 13 chat-999
>>>
>>>
>>>
>>>
>>>
>>>         _______________________________________________
>>>         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://linkedin.com/in/miconda  -- http://twitter.com/miconda
>>
>>
>
>     -- 
>     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

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From abalashov at evaristesys.com  Fri Jun  3 11:33:41 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 03 Jun 2011 05:33:41 -0400
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
In-Reply-To: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>
References: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>
Message-ID: <4DE8AA75.902@evaristesys.com>

This sounds like an asymmetrical engagement of rtpproxy (e.g. on the 
reply but not the initial request, or vice versa).

On 06/02/2011 09:45 PM, Mokhtar Bengana wrote:

> I have Kamailio and rtpproxy running behind NAT on the same server.
> Inbound calls to Kamailio from the outside work fine and I have two
> way audio but when I try an outbound echo test call the call connects
> but I have no audio. rtpproxy seems to be running and all rtp ports
> are poiting to Kamailio internal ip. The syslog is showing this error.
> Not sure what's missing in the config. Appreciate any inputs. Thanks.
>
>
>
> root at Kamailio:~# tail /var/log/syslog
> Jun  3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy
> [rtpproxy.c:1403]: rtp proxy<udp:127.0.0.1:22222>  found, support for
> it enabled
> Jun  3 01:25:05 Kamailio kamailio[791]: INFO:<core>  [forward.c:786]:
> broken reply to forward - no 2nd via
> Jun  3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy
> [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
> Jun  3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy
> [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
>
> _______________________________________________
> 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
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From leroi.lists at gmail.com  Fri Jun  3 12:12:38 2011
From: leroi.lists at gmail.com (Roman Yeryomin)
Date: Fri, 3 Jun 2011 13:12:38 +0300
Subject: [SR-Users] perlvdb error
In-Reply-To: <4DE88A5A.2010206@gmail.com>
References: <BANLkTik+3HUgwA3HGfhfJRNVK_bJycccmw@mail.gmail.com>
	<4DAC09CA.7080206@gmail.com>
	<BANLkTi=BKYYRu1KpfU73kCG21iwnrM==Bg@mail.gmail.com>
	<201104181447.03413.henning.westerholt@1und1.de>
	<BANLkTinrfwujAZSdwS+iBQxStaj5OX7uxg@mail.gmail.com>
	<BANLkTi=NpFr5MRBLnVQ0TfbmB35ELB_VBQ@mail.gmail.com>
	<BANLkTinxK-nj_NY+nszFQigN_txgwWjWkg@mail.gmail.com>
	<4DAED065.2040300@gmail.com>
	<BANLkTinoqO5g=FwngV4CEmo5CJnGQ8Y1xg@mail.gmail.com>
	<4DAFF5F4.1020805@gmail.com>
	<BANLkTin8YzyNqrZ9sHDLsMrK4MATT1tTeA@mail.gmail.com>
	<4DB6891E.4060102@gmail.com>
	<BANLkTimX=bjhP1pgZ+WPo4SBVFBqo2kU1Q@mail.gmail.com>
	<4DB82787.6020401@gmail.com>
	<BANLkTinkxHA2u69WRgafMKL=6=EZHGVCNA@mail.gmail.com>
	<BANLkTinLif8+Gh6hM5MzzrQD6qxRGLWxvQ@mail.gmail.com>
	<4DBEA7DD.7070500@gmail.com>
	<BANLkTikj4zZcqNgVzPhyj9AbZ5B9sy7jsA@mail.gmail.com>
	<4DD4CA9B.3040109@gmail.com> <4DE88A5A.2010206@gmail.com>
Message-ID: <BANLkTikAZjf8LhbRpjxMN5+WvBvrrghO2Q@mail.gmail.com>

On 3 June 2011 10:16, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>>>>
>>> Sorry for disturbing... is there any chance?
>>
>> it is, but I cannot promise a date yet -- too many things to catch up,
>> while testing this is very inconvenient not having a way to troubleshoot.
>
> Not to forget about this one, is it possible to provide me access to a test
> machine where you have such configuration already set? Then I could just
> login and troubleshoot, far more faster than it will take to make myself
> such a testbed.
>

I'm afraid not. At least in the form it is now. As it's not real
machine, it's under virtualbox on my development box in local network.
I can give you the image for sure.
Also I will think maybe I can set it up somewhere to give you an access.
And thanks for remembering about this!

Regards,
Roman


From ozren.lapcevic at gmail.com  Fri Jun  3 13:06:52 2011
From: ozren.lapcevic at gmail.com (Ozren Lapcevic)
Date: Fri, 3 Jun 2011 13:06:52 +0200
Subject: [SR-Users] IMC module - #join to private room crashes kamailio
In-Reply-To: <4DE89E7F.3080008@gmail.com>
References: <BANLkTinHnpQ078hCmV1A3L4UqhtUJVuSSw@mail.gmail.com>
	<4DE64CBE.3030607@gmail.com>
	<BANLkTinDze_Kk4rBbrtwEfy_Ub8WKWFDag@mail.gmail.com>
	<4DE88B14.7040603@gmail.com>
	<BANLkTi=0qzp-qXAQQMS4WJ8pHtu_vLDpVA@mail.gmail.com>
	<4DE89E7F.3080008@gmail.com>
Message-ID: <BANLkTinZ2jDioeYscKG7-=1qKQEqG8DXoA@mail.gmail.com>

Hi,

it works now! Thanks for the quick fix.

Cheers
Ozren


On Fri, Jun 3, 2011 at 10:42 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
> I did a fix for the case of joining private rooms without permissions, can
> you try it again (pull the git branch 3.1)?
>
> Thanks,
> Daniel
>
>
>
> On 6/3/11 9:59 AM, Ozren Lapcevic wrote:
>
> Hi,
>
> I previously got only one core file, but now that I've enabled one core
> file per process, I got two core files. Here are the backtraces:
>
> #0  0x00007fc9c5787ed5 in raise () from /lib/libc.so.6
> #1  0x00007fc9c57893f3 in abort () from /lib/libc.so.6
> #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
> main.c:651
> #3  <signal handler called>
> #4  0x00007fc9c5822307 in syscall () from /lib/libc.so.6
> #5  0x00007fc9c0da50c6 in imc_del_room (name=0x7fc9bea13438,
> domain=0x7fc9bea13448) at ../../mem/../futexlock.h:123
> #6  0x00007fc9c0da51d8 in imc_htable_destroy () at imc_mng.c:104
> #7  0x00007fc9c0d9947a in destroy () at imc.c:736
> #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
> #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
> #10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
> show_status=1) at main.c:668
> #11 0x0000000000463e9c in handle_sigs () at main.c:759
> #12 0x0000000000465572 in main_loop () at main.c:1676
> #13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
> main.c:2398
>
> and:
>
> #0  imc_handle_join (msg=<value optimized out>, cmd=<value optimized out>,
> src=<value optimized out>, dst=<value optimized out>) at imc_cmd.c:329
> #1  0x00007fc9c0d985b0 in imc_manager (msg=0x91ce20, str1=<value optimized
> out>, str2=<value optimized out>) at imc.c:544
> #2  0x000000000041591f in do_action (h=0x7fffffffad30, a=0x925c28,
> msg=0x91ce20) at action.c:1095
> #3  0x000000000041cc5e in run_actions (h=0x7fffffffad30, a=0x925c28,
> msg=0x91ce20) at action.c:1555
> #4  0x000000000041cf97 in run_actions_safe (h=0x7fffffffbdb0,
> a=0x7fc9c5873798, msg=0x0) at action.c:1607
> #5  0x00000000004b1574 in rval_get_int (h=0x7fffffffbdb0, msg=<value
> optimized out>, i=0x7fffffffb408, rv=0xd1d990, cache=0xd1d910) at
> rvalue.c:904
> #6  0x00000000004b4fdc in rval_expr_eval_int (h=0x7fffffffbdb0,
> msg=0x91ce20, res=0x7fffffffb408, rve=0x9262c0) at rvalue.c:1866
> #7  0x0000000000416e02 in do_action (h=0x7fffffffbdb0, a=0x927000,
> msg=0x91ce20) at action.c:1071
> #8  0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x9258a0,
> msg=0x91ce20) at action.c:1555
> #9  0x0000000000416e60 in do_action (h=0x7fffffffbdb0, a=0x927ab8,
> msg=0x91ce20) at action.c:1090
> #10 0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x921c38,
> msg=0x91ce20) at action.c:1555
> #11 0x00000000004177e5 in do_action (h=0x7fffffffbdb0, a=0x90cc40,
> msg=0x91ce20) at action.c:711
> #12 0x000000000041cc5e in run_actions (h=0x7fffffffbdb0, a=0x9079e0,
> msg=0x91ce20) at action.c:1555
> #13 0x000000000041cf23 in run_top_route (a=0x9079e0, msg=0x91ce20, c=<value
> optimized out>) at action.c:1628
> #14 0x00000000004958c6 in receive_msg (
>     buf=0x8a3a00 "MESSAGE sip:chat-999 at abc.hr SIP/2.0\r\nCall-ID:
> 9d5779ef64ae7394e7b922cc4e98e424 at 0.0.0.0\r\nCSeq: 11051797
> MESSAGE\r\nFrom: \"oz\" <sip:oz at abc.hr>;tag=6d7f9acd\r\nTo: <
> sip:chat-999 at abc.hr>\r\nVia: SIP/2.0/UDP 161"..., len=<value optimized
> out>, rcv_info=0x7fffffffc030) at receive.c:205
> #15 0x000000000051c861 in udp_rcv_loop () at udp_server.c:532
> #16 0x0000000000464b15 in main_loop () at main.c:1554
> #17 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
> main.c:2398
>
>
> Best regards,
> Ozren
>
>
> On Fri, Jun 3, 2011 at 9:19 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> did you get only one core file? This one seems from the main process when
>> it discovered that a SIP worked crashed.
>>
>> If you got more than one, send the backtraces from the other core files.
>> If you got only one, please enable one core file per process and re-do it
>> again.
>>
>> See last part of:
>> http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles
>>
>> Cheers,
>>  Daniel
>>
>>
>>
>> On 6/2/11 3:57 PM, Ozren Lapcevic wrote:
>>
>> Hi Daniel,
>>
>> here is the backtrace:
>>
>> #0  0x00007ffc46b81ed5 in raise () from /lib/libc.so.6
>> #1  0x00007ffc46b833f3 in abort () from /lib/libc.so.6
>> #2  0x000000000046119e in sig_alarm_abort (signo=<value optimized out>) at
>> main.c:651
>> #3  <signal handler called>
>> #4  0x00007ffc46c1c307 in syscall () from /lib/libc.so.6
>> #5  0x00007ffc4219f115 in imc_del_room (name=0x7ffc3fe0d388,
>> domain=0x7ffc3fe0d398) at ../../mem/../futexlock.h:110
>> #6  0x00007ffc4219f1d8 in imc_htable_destroy () at imc_mng.c:104
>> #7  0x00007ffc4219347a in destroy () at imc.c:736
>> #8  0x00000000004d9964 in destroy_modules () at sr_module.c:732
>> #9  0x0000000000462930 in cleanup (show_status=1) at main.c:527
>> #10 0x0000000000463564 in shutdown_children (sig=<value optimized out>,
>> show_status=1) at main.c:668
>> #11 0x0000000000463e9c in handle_sigs () at main.c:759
>> #12 0x0000000000465572 in main_loop () at main.c:1676
>> #13 0x0000000000467f53 in main (argc=<value optimized out>, argv=0x0) at
>> main.c:2398
>>
>> I'm not sure if I got the right thing. I got it with gdb
>> /usr/local/kamailio-ozren/sbin/kamailio --core core and called bt in gdb.
>>
>> I've repeated scenario described in the first mail: user tries to join in
>> a private room without getting an invite first.
>>
>> Best regards,
>> Ozren
>>
>>
>> On Wed, Jun 1, 2011 at 4:29 PM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> can you get the core dump and send the backtrace?
>>>
>>> Be sure you have 'ulimit -c unlimited', working directory (-w option) set
>>> to writable path and you don't run it in chowner (no -u or -g parameters).
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 6/1/11 4:25 PM, Ozren Lapcevic wrote:
>>>
>>>  Hi,
>>>
>>> there is a bug in IMC module that crashes Kamailio.
>>>
>>> Here is the scenario: One user (pero at abc.hr) creates private chat room (
>>> chat-999 at abc.hr). Other user (oz at abc.hr) tries to join the
>>> chat-999 at abc.hr by sending #join. Kamailio crashes at that point. Here
>>> is the log:
>>>
>>> Best regards,
>>> Ozren
>>>
>>>
>>> Jun  1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]:
>>> segfault at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in
>>> imc.so[7f28f89cf000+17000]
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>>> INFO: <script>: going to imc: F [sip:oz at abc.hr] T [sip:chat-999 at abc.hr]
>>> B [#join]
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: <core> [receive.c:186]: preparing to run routing scripts...
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>>> DEBUG: imc [imc.c:525]: found command
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: maxfwd [mf_funcs.c:85]: value = 70
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>>> DEBUG: imc [imc_cmd.c:146]: command: [join]
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body :
>>> content_length=223
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>>> DEBUG: imc [imc_cmd.c:292]: found room [chat-999]
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: <core> [parser/msg_parser.c:103]: found end of header
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]:
>>> ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user
>>> [oz]
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>>> ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
>>> DEBUG: <core> [parser/msg_parser.c:630]: SIP Request:
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>>> ALERT: <core> [main.c:744]: core was not generated
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]:
>>> DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82,
>>> 161.53.255.82, 0)
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]:
>>> DEBUG: <core> [parser/msg_parser.c:632]:  method:  <MESSAGE>
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>>> INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]:
>>> INFO: <core> [main.c:807]: INFO: signal 15 received
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>>> DEBUG: imc [imc.c:709]: room 3 chat-777
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>>> DEBUG: imc [imc.c:709]: room 11 chat-888
>>> Jun  1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]:
>>> DEBUG: imc [imc.c:709]: room 13 chat-999
>>>
>>>
>>>
>>>
>>>
>>>  _______________________________________________
>>> 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://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
>
> _______________________________________________
> 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 Mierlahttp://www.asipto.com
>
>
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From mokhtar.bengana at gmail.com  Fri Jun  3 15:33:22 2011
From: mokhtar.bengana at gmail.com (Mokhtar Bengana)
Date: Fri, 3 Jun 2011 08:33:22 -0500
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
In-Reply-To: <4DE8AA75.902@evaristesys.com>
References: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>
	<4DE8AA75.902@evaristesys.com>
Message-ID: <BANLkTinv88kitQE3KmD9+F4ZRiGh+XAysA@mail.gmail.com>

This is how I configured rtpproxy. Not sure why rtpproxy is not
engaged both ways. Thanks for your help.


root at Kamailio:/etc/default# more rtpproxy
# Defaults for rtpproxy

# The control socket.
#CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock"
# To listen on an UDP socket, uncomment this line:

LISTEN_ADDR=public ip address
CONTROL_SOCK="udp:localhost:22222"

# Additional options that are passed to the daemon.
EXTRA_OPTS="-l ${LISTEN_ADDR}"
root at Kamailio:/etc/default#







##kamailio.cfg##



#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif



# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")

# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger at mydomain.com")

# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif




# Caller NAT detection route
route[NAT] {
#!ifdef WITH_NAT
        force_rport();
        if (nat_uac_test("19")) {
                if (method=="REGISTER") {
                        fix_nated_register();
                } else {
                        fix_nated_contact();
                }
                setflag(FLT_NATS);
        }
#!endif
        return;
}




route[RELAY] {
#!ifdef WITH_NAT
        if (check_route_param("nat=yes")) {
                setbflag(FLB_NATB);
        }
        if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
                route(RTPPROXY);
        }
#!endif




# RTPProxy control
route[RTPPROXY] {
#!ifdef WITH_NAT
        if (is_method("BYE")) {
                unforce_rtp_proxy();
        } else if (is_method("INVITE")){
                force_rtp_proxy();
        }
        if (!has_totag()) add_rr_param(";nat=yes");
#!endif
        return;
}












On Fri, Jun 3, 2011 at 4:33 AM, Alex Balashov <abalashov at evaristesys.com> wrote:
> This sounds like an asymmetrical engagement of rtpproxy (e.g. on the reply
> but not the initial request, or vice versa).
>
> On 06/02/2011 09:45 PM, Mokhtar Bengana wrote:
>
>> I have Kamailio and rtpproxy running behind NAT on the same server.
>> Inbound calls to Kamailio from the outside work fine and I have two
>> way audio but when I try an outbound echo test call the call connects
>> but I have no audio. rtpproxy seems to be running and all rtp ports
>> are poiting to Kamailio internal ip. The syslog is showing this error.
>> Not sure what's missing in the config. Appreciate any inputs. Thanks.
>>
>>
>>
>> root at Kamailio:~# tail /var/log/syslog
>> Jun ?3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy
>> [rtpproxy.c:1403]: rtp proxy<udp:127.0.0.1:22222> ?found, support for
>> it enabled
>> Jun ?3 01:25:05 Kamailio kamailio[791]: INFO:<core> ?[forward.c:786]:
>> broken reply to forward - no 2nd via
>> Jun ?3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy
>> [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
>> Jun ?3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy
>> [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
>>
>> _______________________________________________
>> 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
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.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 miconda at gmail.com  Fri Jun  3 16:02:26 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 03 Jun 2011 16:02:26 +0200
Subject: [SR-Users] IRC meeting today, 15:00GMT
In-Reply-To: <4DE889F7.5080505@gmail.com>
References: <4DE889F7.5080505@gmail.com>
Message-ID: <4DE8E972.3060508@gmail.com>

Hello,

probably the gmt is not taking the summer daylight saving time, anyhow, 
we are ready to start the irc meeting now if you want to join.

Cheers,
Daniel

On 6/3/11 9:15 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> everyone that wants to discuss about the next major release (v3.2.0), 
> please join us on IRC:
>
> Channel: #sip-router
> Server: irc.freenode.net
> Time: 15:00GMT/London (16:00 Berlin/Paris/Madrid, 10:00am New York)
>
> Cheers,
> Daniel
>

-- 
Daniel-Constantin Mierla
http://www.asipto.com



From bruno.bresciani at gmail.com  Fri Jun  3 16:21:34 2011
From: bruno.bresciani at gmail.com (Bruno Bresciani)
Date: Fri, 3 Jun 2011 11:21:34 -0300
Subject: [SR-Users] callback function of tm module
In-Reply-To: <4DE89DAD.3060501@1und1.de>
References: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>
	<4DE89DAD.3060501@1und1.de>
Message-ID: <BANLkTik_8Tz=B_gO4-khLdWSe6Pm6TBDAw@mail.gmail.com>

Hi Timo,

In file t_hook.c line 386 "*cbp->callback( trans, cbp->types, params )*"
calls the function to record register_tmcb when I received a INVITE. I'd
like to now how register_tmcb function is call by the function callback.


Regards,




2011/6/3 Timo Reimann <timo.reimann at 1und1.de>

> Hi Bruno,
>
>
> On 03.06.2011 00:07, Bruno Bresciani wrote:
> > I'm having doubts in the implementation of the callback function module
> > tm. As she calls the function to record register_tmcb ()?
> > Can someome help me?
>
> Could you be more specific on which part of the tm module you are having
> trouble with?
>
> If you need an example on how to use tm's callbacks you may take a look
> at the dialog module, specifically the files dialog.c and dlg_handlers.c.
>
>
> Cheers,
>
> --Timo
>
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From timo.reimann at 1und1.de  Fri Jun  3 17:59:39 2011
From: timo.reimann at 1und1.de (Timo Reimann)
Date: Fri, 3 Jun 2011 17:59:39 +0200
Subject: [SR-Users] callback function of tm module
In-Reply-To: <BANLkTik_8Tz=B_gO4-khLdWSe6Pm6TBDAw@mail.gmail.com>
References: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>	<4DE89DAD.3060501@1und1.de>
	<BANLkTik_8Tz=B_gO4-khLdWSe6Pm6TBDAw@mail.gmail.com>
Message-ID: <4DE904EB.9030007@1und1.de>

Hey Bruno,


On 03.06.2011 16:21, Bruno Bresciani wrote:
> In file t_hook.c line 386 "*cbp->callback( trans, cbp->types, params )*"
> calls the function to record register_tmcb when I received a INVITE. I'd
> like to now how register_tmcb function is call by the function callback.

register_tmcb() isn't called by a callback function, it rather works the
other way around. For clarity, here's the call flow:

(1) A user is interested in being called back on a particular tm event,
e.g., TMCB_DESTROY. To be notified of such events, he calls
tmcb_register() and passes the set of events (callback types) he is
interested in (e.g., TMCB_DESTROY and possibly others) together with the
desired callback function and a few other parameters.
(2) In tmcb_register(), a few sanity checks are done first (e.g.,
callback type is valid, callback function is not NULL, transaction
exists, etc.). If they pass, the callback function and the parameters
are stored in a list of callbacks (called cb_list in register_tmcb()).
(3) When a particular tm event occurs, the tm module checks if any
callbacks were registered for that specific event. If so, it executes
each registered callback function in sequence, with each function being
passed the callback type and callback-specific parameter. The call of
the callback is exactly what

  cbp->callback( trans, type, params )

does, at least for call types other than TMCB_REQUEST_IN.

For TMCB_REQUEST_IN, the handling is slightly different because
callbacks for new SIP requests cannot be associated with an already
existing transaction (after all, they are new). With regards to how
TMCB_REQUEST_IN-typed callback functions are called, the only difference
is that the callback function is being passed the set of all callback
types the registering user was interested in, and not just TMCB_REQUEST_IN:

  cbp->callback( trans, cbp->types, params )

(This is line 386.) Actually, I am not quite sure why the extent of
returned callback types differs here. I'm using tm callbacks myself but
never had to take advantage of that.


Anyways, if all you want to do is use tm callbacks from a module of
yours, just call register_tmcb() passing

- a callback function,
- a transaction cell (unless you're registering for TMCB_REQUEST_IN),
- a void pointer to something you want to be passed back on callback
execution, and
- an optional release function for cleanup purposes.

Implement your callback-specific logic in the callback function, and
that's it. Naturally, your callback function must fit the callback
signature defined in t_hooks.h.


I hope this answers your question. If not, let us know what you
specifically have in mind, i.e., whether you would like to use tm
callbacks, change the framework, or whatever.


Cheers,

--Timo



> 2011/6/3 Timo Reimann <timo.reimann at 1und1.de <mailto:timo.reimann at 1und1.de>>
> 
>     Hi Bruno,
> 
> 
>     On 03.06.2011 00:07, Bruno Bresciani wrote:
>     > I'm having doubts in the implementation of the callback function
>     module
>     > tm. As she calls the function to record register_tmcb ()?
>     > Can someome help me?
> 
>     Could you be more specific on which part of the tm module you are having
>     trouble with?
> 
>     If you need an example on how to use tm's callbacks you may take a look
>     at the dialog module, specifically the files dialog.c and
>     dlg_handlers.c.
> 
> 
>     Cheers,
> 
>     --Timo


From bruno.bresciani at gmail.com  Fri Jun  3 20:00:19 2011
From: bruno.bresciani at gmail.com (Bruno Bresciani)
Date: Fri, 3 Jun 2011 15:00:19 -0300
Subject: [SR-Users] callback function of tm module
In-Reply-To: <4DE904EB.9030007@1und1.de>
References: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>
	<4DE89DAD.3060501@1und1.de>
	<BANLkTik_8Tz=B_gO4-khLdWSe6Pm6TBDAw@mail.gmail.com>
	<4DE904EB.9030007@1und1.de>
Message-ID: <BANLkTikEUqLHSK4cD4AK=nSC43mFKsY9KQ@mail.gmail.com>

Actually I'm migrating my call routing module of version 1.5.0 of kamailio
to version 3.1.2. I want to understand why the tm module registers an event
callback after the INVITE message is processed by my routing module.
When I use the dispatcher module for routing, no event callback is
registered by the tm module.

I do not quite understand this process callback, I study a little more ...


Regards,



2011/6/3 Timo Reimann <timo.reimann at 1und1.de>

> Hey Bruno,
>
>
> On 03.06.2011 16:21, Bruno Bresciani wrote:
> > In file t_hook.c line 386 "*cbp->callback( trans, cbp->types, params )*"
> > calls the function to record register_tmcb when I received a INVITE. I'd
> > like to now how register_tmcb function is call by the function callback.
>
> register_tmcb() isn't called by a callback function, it rather works the
> other way around. For clarity, here's the call flow:
>
> (1) A user is interested in being called back on a particular tm event,
> e.g., TMCB_DESTROY. To be notified of such events, he calls
> tmcb_register() and passes the set of events (callback types) he is
> interested in (e.g., TMCB_DESTROY and possibly others) together with the
> desired callback function and a few other parameters.
> (2) In tmcb_register(), a few sanity checks are done first (e.g.,
> callback type is valid, callback function is not NULL, transaction
> exists, etc.). If they pass, the callback function and the parameters
> are stored in a list of callbacks (called cb_list in register_tmcb()).
> (3) When a particular tm event occurs, the tm module checks if any
> callbacks were registered for that specific event. If so, it executes
> each registered callback function in sequence, with each function being
> passed the callback type and callback-specific parameter. The call of
> the callback is exactly what
>
>  cbp->callback( trans, type, params )
>
> does, at least for call types other than TMCB_REQUEST_IN.
>
> For TMCB_REQUEST_IN, the handling is slightly different because
> callbacks for new SIP requests cannot be associated with an already
> existing transaction (after all, they are new). With regards to how
> TMCB_REQUEST_IN-typed callback functions are called, the only difference
> is that the callback function is being passed the set of all callback
> types the registering user was interested in, and not just TMCB_REQUEST_IN:
>
>  cbp->callback( trans, cbp->types, params )
>
> (This is line 386.) Actually, I am not quite sure why the extent of
> returned callback types differs here. I'm using tm callbacks myself but
> never had to take advantage of that.
>
>
> Anyways, if all you want to do is use tm callbacks from a module of
> yours, just call register_tmcb() passing
>
> - a callback function,
> - a transaction cell (unless you're registering for TMCB_REQUEST_IN),
> - a void pointer to something you want to be passed back on callback
> execution, and
> - an optional release function for cleanup purposes.
>
> Implement your callback-specific logic in the callback function, and
> that's it. Naturally, your callback function must fit the callback
> signature defined in t_hooks.h.
>
>
> I hope this answers your question. If not, let us know what you
> specifically have in mind, i.e., whether you would like to use tm
> callbacks, change the framework, or whatever.
>
>
> Cheers,
>
> --Timo
>
>
>
> > 2011/6/3 Timo Reimann <timo.reimann at 1und1.de <mailto:
> timo.reimann at 1und1.de>>
> >
> >     Hi Bruno,
> >
> >
> >     On 03.06.2011 00:07, Bruno Bresciani wrote:
> >     > I'm having doubts in the implementation of the callback function
> >     module
> >     > tm. As she calls the function to record register_tmcb ()?
> >     > Can someome help me?
> >
> >     Could you be more specific on which part of the tm module you are
> having
> >     trouble with?
> >
> >     If you need an example on how to use tm's callbacks you may take a
> look
> >     at the dialog module, specifically the files dialog.c and
> >     dlg_handlers.c.
> >
> >
> >     Cheers,
> >
> >     --Timo
>
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From gchen00 at insightbb.com  Fri Jun  3 21:58:24 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 3 Jun 2011 15:58:24 -0400 (EDT)
Subject: [SR-Users] Questino about dialplan module
In-Reply-To: <680529710.278782.1307128446251.JavaMail.root@md09.insight.synacor.com>
Message-ID: <738223519.279719.1307131104796.JavaMail.root@md09.insight.synacor.com>

Kamailio 3.1.2 


I am testing out dialplan module. Every time when I issue 'kamctl dialplan show', it shows what is in the database even if I have not yet issue "kamctl dialplan reload' after I changed data in database. I thought that dialplan is stored in the memory and you have to issue the reload before you can see the change. Why the ' kamctl dialplan show' display the data directly from mysql database? Does that mean that dialplan data is not stored in the memory? 


Gary 
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From jh at tutpro.com  Fri Jun  3 21:49:37 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Fri, 3 Jun 2011 22:49:37 +0300
Subject: [SR-Users]  Questino about dialplan module
In-Reply-To: <738223519.279719.1307131104796.JavaMail.root@md09.insight.synacor.com>
References: <680529710.278782.1307128446251.JavaMail.root@md09.insight.synacor.com>
	<738223519.279719.1307131104796.JavaMail.root@md09.insight.synacor.com>
Message-ID: <19945.15057.604936.302312@sip.test.fi>

Gary Chen writes:

> I am testing out dialplan module. Every time when I issue 'kamctl
> dialplan show', it shows what is in the database even if I have not
> yet issue "kamctl dialplan reload' after I changed data in database. I
> thought that dialplan is stored in the memory and you have to issue
> the reload before you can see the change. Why the ' kamctl dialplan
> show' display the data directly from mysql database? Does that mean
> that dialplan data is not stored in the memory?  

there is no mi function to show dialplan rules in memory.  i have no
idea, what 'show' does.  please read readme of dialplan module before
asking questions about the module.

-- juha


From gchen00 at insightbb.com  Fri Jun  3 23:05:51 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Fri, 3 Jun 2011 17:05:51 -0400 (EDT)
Subject: [SR-Users] Questino about dialplan module
In-Reply-To: <1472446096.281056.1307135000796.JavaMail.root@md09.insight.synacor.com>
Message-ID: <1249927071.281112.1307135151953.JavaMail.root@md09.insight.synacor.com>



I am talking about kamctl dialplan command not mi. 

When you type in kamctl, it lists a lot commands. Are these commands still valid? 

See below: 

kamctl 



../../sbin/kamctl 3.1.0 




Existing commands: 




-- command 'start|stop|restart' 




restart ............................ restart Kamailio 

start .............................. start Kamailio 

stop ............................... stop Kamailio 




-- command 'acl' - manage access control lists (acl) 




acl show [<username>] .............. show user membership 

acl grant <username> <group> ....... grant user membership (*) 

acl revoke <username> [<group>] .... grant user membership(s) (*) 




-- command 'lcr' - manage least cost routes (lcr) 




lcr show_gws....... show database gateways 

lcr show_routes.... show database routes 

lcr dump_gws....... show in memory gateways 

lcr dump_routes.... show in memory routes 

lcr reload ........ reload lcr gateways and routes 

lcr eval_weights .. evaluates probability for given GW's weights 




-- command 'cr' - manage carrierroute tables 




cr show ....................................................... show tables 

cr reload ..................................................... reload tables 

cr dump ....................................................... show in memory tables 

cr addcn <carrier id> <carrier name> .......................... add a carrier name 

cr rmcn <carrier id> ......................................... rm a carrier name 

cr adddn <domain id> <domain name> ............................ add a domain name 

cr rmdn <domain id> .......................................... rm a domain name 

cr addcarrier <carrier> <scan_prefix> <domain> <rewrite_host> ................ 

<prob> <strip> <rewrite_prefix> <rewrite_suffix> ............... 

<flags> <mask> <comment> .........................add a carrier 

(prob, strip, rewrite_prefix, rewrite_suffix,................... 

flags, mask and comment are optional arguments) ............... 

cr rmcarrier <carrier> <scan_prefix> <domain> ................ rm a carrier 




-- command 'rpid' - manage Remote-Party-ID (RPID) 




rpid add <username> <rpid> ......... add rpid for a user (*) 

rpid rm <username> ................. set rpid to NULL for a user (*) 

rpid show <username> ............... show rpid of a user 




-- command 'add|passwd|rm' - manage subscribers 




add <username> <password> .......... add a new subscriber (*) 

passwd <username> <passwd> ......... change user's password (*) 

rm <username> ...................... delete a user (*) 




-- command 'add|dump|reload|rm|show' - manage trusted 




trusted show ...................... show db content 

trusted dump ...................... show cache content 

trusted reload .................... reload db table into cache 

trusted add <src_ip> <proto> <from_pattern> <tag> 

....................... add a new entry 

....................... (from_pattern and tag are optional arguments) 

trusted rm <src_ip> ............... remove all entres for the given src_ip 




-- command 'dispatcher' - manage dispatcher 




* Examples: dispatcher addgw 1 sip:1.2.3.1:5050 1 'outbound gateway' 

* dispatcher addgw 2 sip:1.2.3.4:5050 3 '' 

* dispatcher rmgw 4 

dispatcher show ..................... show dispatcher gateways 

dispatcher reload ................... reload dispatcher gateways 

dispatcher dump ..................... show in memory dispatcher gateways 

dispatcher addgw <setid> <destination> <flags> <description> 

.......................... add gateway 

dispatcher rmgw <id> ................ delete gateway 




-- command 'db' - database operations 




db exec <query> ..................... execute SQL query 

db roexec <roquery> ................. execute read-only SQL query 

db run <id> ......................... execute SQL query from $id variable 

db rorun <id> ....................... execute read-only SQL query from 

$id variable 

db show <table> ..................... display table content 




-- command 'speeddial' - manage speed dials (short numbers) 




speeddial show <speeddial-id> ....... show speeddial details 

speeddial list <sip-id> ............. list speeddial for uri 

speeddial add <sip-id> <sd-id> <new-uri> [<desc>] ... 

........................... add a speedial (*) 

speeddial rm <sip-id> <sd-id> ....... remove a speeddial (*) 

speeddial help ...................... help message 

- <speeddial-id>, <sd-id> must be an AoR (username at domain) 

- <sip-id> must be an AoR (username at domain) 

- <new-uri> must be a SIP AoR (sip:username at domain) 

- <desc> a description for speeddial 




-- command 'avp' - manage AVPs 




avp list [-T table] [-u <sip-id|uuid>] 

[-a attribute] [-v value] [-t type] ... list AVPs 

avp add [-T table] <sip-id|uuid> 

<attribute> <type> <value> ............ add AVP (*) 

avp rm [-T table] [-u <sip-id|uuid>] 

[-a attribute] [-v value] [-t type] ... remove AVP (*) 

avp help .................................. help message 

- -T - table name 

- -u - SIP id or unique id 

- -a - AVP name 

- -v - AVP value 

- -t - AVP name and type (0 (str:str), 1 (str:int), 

2 (int:str), 3 (int:int)) 

- <sip-id> must be an AoR (username at domain) 

- <uuid> must be a string but not AoR 




-- command 'alias_db' - manage database aliases 




alias_db show <alias> .............. show alias details 

alias_db list <sip-id> ............. list aliases for uri 

alias_db add <alias> <sip-id> ...... add an alias (*) 

alias_db rm <alias> ................ remove an alias (*) 

alias_db help ...................... help message 

- <alias> must be an AoR (username at domain)" 

- <sip-id> must be an AoR (username at domain)" 




-- command 'domain' - manage local domains 




domain reload ....................... reload domains from disk 

domain show ......................... show current domains in memory 

domain showdb ....................... show domains in the database 

domain add <domain> ................. add the domain to the database 

domain rm <domain> .................. delete the domain from the database 




-- command 'cisco_restart' - restart CISCO phone (NOTIFY) 




cisco_restart <uri> ................ restart phone configured for <uri> 




-- command 'online' - dump online users from memory 




online ............................. display online users 




-- command 'monitor' - show internal status 




monitor ............................ show server's internal status 




-- command 'ping' - ping a SIP URI (OPTIONS) 




ping <uri> ......................... ping <uri> with SIP OPTIONS 




-- command 'ul|alias' - manage user location or aliases 




ul show [<username>]................ show in-RAM online users 

ul show --brief..................... show in-RAM online users in short format 

ul rm <username> [<contact URI>].... delete user's usrloc entries 

ul add <username> <uri> ............ introduce a permanent usrloc entry 

ul add <username> <uri> <expires> .. introduce a temporary usrloc entry 




-- command 'fifo' 




fifo ............................... send raw FIFO command 




-- command 'cisco_restart' - restart CISCO phone (NOTIFY) 




cisco_restart <uri> ................ restart phone configured for <uri> 




-- command 'online' - dump online users from memory 




online ............................. display online users 




-- command 'monitor' - show internal status 




monitor ............................ show server's internal status 




-- command 'ping' - ping a SIP URI (OPTIONS) 




ping <uri> ......................... ping <uri> with SIP OPTIONS 




-- command 'ul|alias' - manage user location or aliases 




ul show [<username>]................ show in-RAM online users 

ul show --brief..................... show in-RAM online users in short format 

ul rm <username> [<contact URI>].... delete user's usrloc entries 

ul add <username> <uri> ............ introduce a permanent usrloc entry 

ul add <username> <uri> <expires> .. introduce a temporary usrloc entry 




-----Original Message----- 
From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Juha Heinanen 
Sent: Friday, June 03, 2011 3:50 PM 
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List 
Subject: [SR-Users] Questino about dialplan module 



Gary Chen writes: 



> I am testing out dialplan module. Every time when I issue 'kamctl 

> dialplan show', it shows what is in the database even if I have not 

> yet issue "kamctl dialplan reload' after I changed data in database. I 

> thought that dialplan is stored in the memory and you have to issue 

> the reload before you can see the change. Why the ' kamctl dialplan 

> show' display the data directly from mysql database? Does that mean 

> that dialplan data is not stored in the memory? 



there is no mi function to show dialplan rules in memory. i have no 

idea, what 'show' does. please read readme of dialplan module before 

asking questions about the module. 



-- juha 



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From Eugen.Dedu at pu-pm.univ-fcomte.fr  Sun Jun  5 22:31:03 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Sun, 05 Jun 2011 22:31:03 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
Message-ID: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>

Hi,

ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for 
users who are not registered an empty NOTIFY body is returned when asked 
by a SUBSCRIBE.  What does this mean from SIP standard point of view, 
and from kamailio point of view (are they identical?)  I see in 
RFC3265/3.1.6.2:
    ....  If the resource
    has no meaningful state at the time that the SUBSCRIBE message is
    processed, this NOTIFY message MAY contain an empty or neutral body
but is difficult for me to interpret what it means.

Example: I ask the presence for a user xyz who registered and quit 
application long time ago:

SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
CSeq: 1 SUBSCRIBE
Via: SIP/2.0/UDP 
82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport
User-Agent: Ekiga/3.3.1
From: <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
Supported: eventlist
To: <sip:xyz at ekiga.net>
Accept: application/pidf+xml
Accept: multipart/related
Accept: application/rlmi+xml
Contact: <sip:eugen.dedu at 82.238.108.175:5060>
Allow: 
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 300
Event: presence
Content-Length: 0
Max-Forwards: 70

I receive the following answer:

NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
CSeq: 1 NOTIFY
Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
User-Agent: Kamailio (1.5.3-notls (i386/linux))
From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
Contact: <sip:86.64.162.35:5060>
Subscription-State: active;expires=370
Event: presence
Content-Length: 0
Max-Forwards: 70

To resume: What does SIP standard say about this NOTIFY with empty body? 
  Does this mean that the user xyz is offline?

Or does this mean that user's status has not changed?  In fact, the 
NOTIFY with empty body (as shown above) is the first one sent by 
kamailio, so there is no "previous" state of that user, hence 
"unchanged" status has no meaning.

Thank you,
-- 
Eugen


From craig at southeren.com  Sun Jun  5 23:21:52 2011
From: craig at southeren.com (Craig Southeren)
Date: Sun, 05 Jun 2011 14:21:52 -0700
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
Message-ID: <4DEBF370.6030302@southeren.com>

On 5/06/2011 1:31 PM, Eugen Dedu wrote:
> Hi,
>
> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for 
> users who are not registered an empty NOTIFY body is returned when 
> asked by a SUBSCRIBE.  What does this mean from SIP standard point of 
> view, and from kamailio point of view (are they identical?)  I see in 
> RFC3265/3.1.6.2:
>    ....  If the resource
>    has no meaningful state at the time that the SUBSCRIBE message is
>    processed, this NOTIFY message MAY contain an empty or neutral body
> but is difficult for me to interpret what it means.
>
> Example: I ask the presence for a user xyz who registered and quit 
> application long time ago:
>
..deleted

> To resume: What does SIP standard say about this NOTIFY with empty 
> body?  Does this mean that the user xyz is offline?
>
> Or does this mean that user's status has not changed?  In fact, the 
> NOTIFY with empty body (as shown above) is the first one sent by 
> kamailio, so there is no "previous" state of that user, hence 
> "unchanged" status has no meaning.

Here is my understanding.

If the NOTIFY is the first NOTIFY sent in response to the SUBSCRIBE, 
then the NOTIFY should be ignored. Opal should be doing this.

if the NOTIFY is *not* the first NOTIFY, i.e. there was a previous 
NOTIFY that contains a valid presence status, then the previous status 
should be used.

    Craig

-- 

-----------------------------------------------------------------------
  Craig Southeren          Post Increment ? VoIP Consulting and Software
  craigs at postincrement.com.au                   www.postincrement.com.au

  Phone:  +61 243654666      ICQ: #86852844
  Fax:    +61 243656905      MSN: craig_southeren at hotmail.com
  Mobile: +61 417231046      Jabber: craigs at jabber.org

  "Science is the poetry of reality." Richard Dawkins



From robertj at voxlucida.com.au  Mon Jun  6 04:09:29 2011
From: robertj at voxlucida.com.au (Robert Jongbloed)
Date: Mon, 6 Jun 2011 12:09:29 +1000
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DEBF370.6030302@southeren.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEBF370.6030302@southeren.com>
Message-ID: <000a01cc23ee$c720b5a0$556220e0$@com.au>

> -----Original Message-----
> From: Craig Southeren [mailto:craig at southeren.com]
> Sent: Monday, 6 June 2011 7:22 AM
...
> > To resume: What does SIP standard say about this NOTIFY with empty
> > body?  Does this mean that the user xyz is offline?

The specification gives no meaning to an empty body. "No meaning" does not mean "offline". Offline means something!

The only purpose of sending a NOTIFY with an empty body, as per the specification, is to finalise the establishment of a subscription when you do not yet have any state information. Any other use is "undocumented", so there had better be a large body of evidence that "everyone does it that way" before we use any semantics.

> > Or does this mean that user's status has not changed?  In fact, the
> > NOTIFY with empty body (as shown above) is the first one sent by
> > kamailio, so there is no "previous" state of that user, hence
> > "unchanged" status has no meaning.
> 
> Here is my understanding.
> 
> If the NOTIFY is the first NOTIFY sent in response to the SUBSCRIBE,
> then the NOTIFY should be ignored. Opal should be doing this.
> 
> if the NOTIFY is *not* the first NOTIFY, i.e. there was a previous
> NOTIFY that contains a valid presence status, then the previous status
> should be used.

Interestingly, I was doing some Googling on this and it appears that the first notify, when it is empty, is supposed to have a Subscription-State header field value of "pending". Again the specification is not clear but there were some discussion groups that mentioned this.

Unfortunately, Kamailio does not do this. So we could not even use this as a discriminator.


Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.




From ibc at aliax.net  Mon Jun  6 10:25:42 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 10:25:42 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest match
Message-ID: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>

Hi, I wonder how feasible is the folowing:

I have a regular expresion like:

  REGEXP = "^((1|20)|(13|3|4)|(5|6))"

There are 3 matching groups:

 1)   (1|20)
 2)   (13|3|4)
 3)   (5|6)

I want to match the RURI user, let's suppose $rU = "138787686223". In
this case the above regexp matches groups 1 and 2, but since group 2
matching is longest ("13" is longest than "1") I want to get value 2.

In case $rU = "999999" I need to get "null", in case $rU =
"1000022323" I need to get "1".

AFAIK there is no way to do it with current regular expression utils
in Kamailio, but maybe I miss something. Thanks a lot.


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


From miconda at gmail.com  Mon Jun  6 10:32:26 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 10:32:26 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
 match
In-Reply-To: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
Message-ID: <4DEC909A.8010506@gmail.com>

Hello,

On 6/6/11 10:25 AM, I?aki Baz Castillo wrote:
> Hi, I wonder how feasible is the folowing:
>
> I have a regular expresion like:
>
>    REGEXP = "^((1|20)|(13|3|4)|(5|6))"
>
> There are 3 matching groups:
>
>   1)   (1|20)
>   2)   (13|3|4)
>   3)   (5|6)
>
> I want to match the RURI user, let's suppose $rU = "138787686223". In
> this case the above regexp matches groups 1 and 2, but since group 2
> matching is longest ("13" is longest than "1") I want to get value 2.
>
> In case $rU = "999999" I need to get "null", in case $rU =
> "1000022323" I need to get "1".
>
> AFAIK there is no way to do it with current regular expression utils
> in Kamailio, but maybe I miss something. Thanks a lot.
one option that comes in my mind is to break it in several if 
conditions, matching one group at a time, written in the order of 
preferences.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Mon Jun  6 10:39:01 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 10:39:01 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <4DEC909A.8010506@gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<4DEC909A.8010506@gmail.com>
Message-ID: <BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>

2011/6/6 Daniel-Constantin Mierla <miconda at gmail.com>:
> one option that comes in my mind is to break it in several if conditions,
> matching one group at a time, written in the order of preferences.

The problem is that it would require perform N regular expression
matching and such expressions will be very long (all the country
prefixes in the world). The main problem is that some groups would
contain prefixes starting with, for example, 12, while in other group
there is the prefix 1 (an the very same in multiple cases, so doing it
manually would be too much manual work and any change in the world
would require hard modifications).

I'm evaluating building a custom function for this purpose.

Thanks.

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


From jh at tutpro.com  Mon Jun  6 10:49:58 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Mon, 6 Jun 2011 11:49:58 +0300
Subject: [SR-Users] Using =~ to get the group identifier with
	longest	match
In-Reply-To: <BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<4DEC909A.8010506@gmail.com>
	<BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
Message-ID: <19948.38070.936143.745283@sip.test.fi>

I?aki Baz Castillo writes:

> The problem is that it would require perform N regular expression
> matching and such expressions will be very long (all the country
> prefixes in the world). The main problem is that some groups would
> contain prefixes starting with, for example, 12, while in other group
> there is the prefix 1 (an the very same in multiple cases, so doing it
> manually would be too much manual work and any change in the world
> would require hard modifications).

perhaps you could use dialplan module.  it there is overlapping regular
expressions, one is selected based on its priority.

-- juha


From miconda at gmail.com  Mon Jun  6 10:53:35 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 10:53:35 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
 match
In-Reply-To: <BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<4DEC909A.8010506@gmail.com>
	<BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
Message-ID: <4DEC958F.3090406@gmail.com>

Hello,

On 6/6/11 10:39 AM, I?aki Baz Castillo wrote:
> 2011/6/6 Daniel-Constantin Mierla<miconda at gmail.com>:
>> one option that comes in my mind is to break it in several if conditions,
>> matching one group at a time, written in the order of preferences.
> The problem is that it would require perform N regular expression
> matching and such expressions will be very long (all the country
> prefixes in the world). The main problem is that some groups would
> contain prefixes starting with, for example, 12, while in other group
> there is the prefix 1 (an the very same in multiple cases, so doing it
> manually would be too much manual work and any change in the world
> would require hard modifications).
>
> I'm evaluating building a custom function for this purpose.
maybe dialplan module could give you a solution -- you can play with 
priorities for matching rules.

Cheers,
Daniel
-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Mon Jun  6 11:00:28 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 11:00:28 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <19948.38070.936143.745283@sip.test.fi>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<4DEC909A.8010506@gmail.com>
	<BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
	<19948.38070.936143.745283@sip.test.fi>
Message-ID: <BANLkTi=J4EDUp9Tjs9mUsfXCdvOXuyXnDw@mail.gmail.com>

2011/6/6 Juha Heinanen <jh at tutpro.com>:
> perhaps you could use dialplan module. ?it there is overlapping regular
> expressions, one is selected based on its priority.

Thanks. I don't need different priorities, I just need that, in case
of multiple matching (i.e. ^1 ,  ^10) the longest matching wins. Does
the dialplan module allow it? By reading the doc it seems that it's
not possible, so I should need to assign lower priority to those
shorter country prefixes. It could be a solution however.

Thanks to both.

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


From jh at tutpro.com  Mon Jun  6 11:03:33 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Mon, 6 Jun 2011 12:03:33 +0300
Subject: [SR-Users] Using =~ to get the group identifier with
	longest	match
In-Reply-To: <BANLkTi=J4EDUp9Tjs9mUsfXCdvOXuyXnDw@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<4DEC909A.8010506@gmail.com>
	<BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
	<19948.38070.936143.745283@sip.test.fi>
	<BANLkTi=J4EDUp9Tjs9mUsfXCdvOXuyXnDw@mail.gmail.com>
Message-ID: <19948.38885.149957.642285@sip.test.fi>

I?aki Baz Castillo writes:

> Thanks. I don't need different priorities, I just need that, in case
> of multiple matching (i.e. ^1 ,  ^10) the longest matching wins. Does
> the dialplan module allow it?

try by giving rule ^1 lower priority than ^10.

-- juha


From ibc at aliax.net  Mon Jun  6 11:12:49 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 11:12:49 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <19948.38885.149957.642285@sip.test.fi>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<4DEC909A.8010506@gmail.com>
	<BANLkTi=5-k2ZTbRHKoZCvzWQCMN73EvKjQ@mail.gmail.com>
	<19948.38070.936143.745283@sip.test.fi>
	<BANLkTi=J4EDUp9Tjs9mUsfXCdvOXuyXnDw@mail.gmail.com>
	<19948.38885.149957.642285@sip.test.fi>
Message-ID: <BANLkTim0t2L8kQzeGRC2PZc-jyKNE3imeQ@mail.gmail.com>

2011/6/6 Juha Heinanen <jh at tutpro.com>:
>> Thanks. I don't need different priorities, I just need that, in case
>> of multiple matching (i.e. ^1 , ?^10) the longest matching wins. Does
>> the dialplan module allow it?
>
> try by giving rule ^1 lower priority than ^10.

Yes, the trick will be:

  priority = 20 - strlen(match_exp)

so longest expressions would have higher priority.

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


From andrew.pogrebennyk at portaone.com  Mon Jun  6 12:37:20 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Mon, 06 Jun 2011 13:37:20 +0300
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
In-Reply-To: <BANLkTinv88kitQE3KmD9+F4ZRiGh+XAysA@mail.gmail.com>
References: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>	<4DE8AA75.902@evaristesys.com>
	<BANLkTinv88kitQE3KmD9+F4ZRiGh+XAysA@mail.gmail.com>
Message-ID: <4DECADE0.3000708@portaone.com>

Mokhtar, could you please make sure that you are calling route(RTPPROXY) 
in reply route as well, as Alex suggested?

On 03.06.2011 16:33, Mokhtar Bengana wrote:
> This is how I configured rtpproxy. Not sure why rtpproxy is not
> engaged both ways. Thanks for your help.

-- 
Sincerely,
Andrew Pogrebennyk


From andrew.pogrebennyk at portaone.com  Mon Jun  6 12:40:30 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Mon, 06 Jun 2011 13:40:30 +0300
Subject: [SR-Users] Questino about dialplan module
In-Reply-To: <738223519.279719.1307131104796.JavaMail.root@md09.insight.synacor.com>
References: <738223519.279719.1307131104796.JavaMail.root@md09.insight.synacor.com>
Message-ID: <4DECAE9E.2040706@portaone.com>

On 03.06.2011 22:58, Gary Chen wrote:
> reload before you can see the change. Why the ' kamctl dialplan show'
> display the data directly from mysql database? Does that mean that
> dialplan data is not stored in the memory?

Gary, a quick look at kamctl script proves that it displays the data 
directly from the mysql database, not from memory:

> ##### ------------------------------------------------ #####
> ### DIALPLAN management
> #
> dialplan() {
>         require_dbengine
>         require_ctlengine
>         case $1 in
>                 show)
>                         shift
>                         if [ $# -gt 0 ] ; then
>                                 mecho "dialplan $1 tables"
>                                 QUERY="select * FROM $DIALPLAN_TABLE WHERE $DIALPLAN_DPID_COLUMN=$1 ORDER BY $DIALPLAN_PR_COLUMN ; "
>                         else
>                                 mecho "dialplan tables"
>                                 QUERY="select * FROM $DIALPLAN_TABLE ORDER BY $DIALPLAN_DPID_COLUMN, $DIALPLAN_PR_COLUMN; "
>                         fi
>                         $DBROCMD "$QUERY"
>                         ;;

-- 
Sincerely,
Andrew Pogrebennyk


From timo.reimann at 1und1.de  Mon Jun  6 12:40:07 2011
From: timo.reimann at 1und1.de (Timo Reimann)
Date: Mon, 6 Jun 2011 12:40:07 +0200
Subject: [SR-Users] callback function of tm module
In-Reply-To: <BANLkTikEUqLHSK4cD4AK=nSC43mFKsY9KQ@mail.gmail.com>
References: <BANLkTinRXdMxfm1WXMYdvV51H80JJp6Rvg@mail.gmail.com>	<4DE89DAD.3060501@1und1.de>	<BANLkTik_8Tz=B_gO4-khLdWSe6Pm6TBDAw@mail.gmail.com>	<4DE904EB.9030007@1und1.de>
	<BANLkTikEUqLHSK4cD4AK=nSC43mFKsY9KQ@mail.gmail.com>
Message-ID: <4DECAE87.5090508@1und1.de>

Hi Bruno,


On 03.06.2011 20:00, Bruno Bresciani wrote:
> Actually I'm migrating my call routing module of version 1.5.0 of
> kamailio to version 3.1.2. I want to understand why the tm module
> registers an event callback after the INVITE message is processed by my
> routing module.
> When I use the dispatcher module for routing, no event callback is
> registered by the tm module.

I am certain that the tm module isn't doing any callback registrations
itself -- at least not by calling register_tmcb() which is used by other
modules exclusively. Some grep'ing and cscope'ing acknowledges that.

It seems to me that some other, intermediary module registers a callback
instead. If it's not a problem for you, run SR in gdb and post a
backtrace from the instance the callback you are wondering about is
allegedly registered. That way, we can take a look at the call flow and
figure out how things work.


Cheers,

--Timo



> 2011/6/3 Timo Reimann <timo.reimann at 1und1.de <mailto:timo.reimann at 1und1.de>>
> 
>     Hey Bruno,
> 
> 
>     On 03.06.2011 16:21, Bruno Bresciani wrote:
>     > In file t_hook.c line 386 "*cbp->callback( trans, cbp->types,
>     params )*"
>     > calls the function to record register_tmcb when I received a
>     INVITE. I'd
>     > like to now how register_tmcb function is call by the function
>     callback.
> 
>     register_tmcb() isn't called by a callback function, it rather works the
>     other way around. For clarity, here's the call flow:
> 
>     (1) A user is interested in being called back on a particular tm event,
>     e.g., TMCB_DESTROY. To be notified of such events, he calls
>     tmcb_register() and passes the set of events (callback types) he is
>     interested in (e.g., TMCB_DESTROY and possibly others) together with the
>     desired callback function and a few other parameters.
>     (2) In tmcb_register(), a few sanity checks are done first (e.g.,
>     callback type is valid, callback function is not NULL, transaction
>     exists, etc.). If they pass, the callback function and the parameters
>     are stored in a list of callbacks (called cb_list in register_tmcb()).
>     (3) When a particular tm event occurs, the tm module checks if any
>     callbacks were registered for that specific event. If so, it executes
>     each registered callback function in sequence, with each function being
>     passed the callback type and callback-specific parameter. The call of
>     the callback is exactly what
> 
>      cbp->callback( trans, type, params )
> 
>     does, at least for call types other than TMCB_REQUEST_IN.
> 
>     For TMCB_REQUEST_IN, the handling is slightly different because
>     callbacks for new SIP requests cannot be associated with an already
>     existing transaction (after all, they are new). With regards to how
>     TMCB_REQUEST_IN-typed callback functions are called, the only difference
>     is that the callback function is being passed the set of all callback
>     types the registering user was interested in, and not just
>     TMCB_REQUEST_IN:
> 
>      cbp->callback( trans, cbp->types, params )
> 
>     (This is line 386.) Actually, I am not quite sure why the extent of
>     returned callback types differs here. I'm using tm callbacks myself but
>     never had to take advantage of that.
> 
> 
>     Anyways, if all you want to do is use tm callbacks from a module of
>     yours, just call register_tmcb() passing
> 
>     - a callback function,
>     - a transaction cell (unless you're registering for TMCB_REQUEST_IN),
>     - a void pointer to something you want to be passed back on callback
>     execution, and
>     - an optional release function for cleanup purposes.
> 
>     Implement your callback-specific logic in the callback function, and
>     that's it. Naturally, your callback function must fit the callback
>     signature defined in t_hooks.h.
> 
> 
>     I hope this answers your question. If not, let us know what you
>     specifically have in mind, i.e., whether you would like to use tm
>     callbacks, change the framework, or whatever.
> 
> 
>     Cheers,
> 
>     --Timo
> 
> 
> 
>     > 2011/6/3 Timo Reimann <timo.reimann at 1und1.de
>     <mailto:timo.reimann at 1und1.de> <mailto:timo.reimann at 1und1.de
>     <mailto:timo.reimann at 1und1.de>>>
>     >
>     >     Hi Bruno,
>     >
>     >
>     >     On 03.06.2011 00:07, Bruno Bresciani wrote:
>     >     > I'm having doubts in the implementation of the callback function
>     >     module
>     >     > tm. As she calls the function to record register_tmcb ()?
>     >     > Can someome help me?
>     >
>     >     Could you be more specific on which part of the tm module you
>     are having
>     >     trouble with?
>     >
>     >     If you need an example on how to use tm's callbacks you may
>     take a look
>     >     at the dialog module, specifically the files dialog.c and
>     >     dlg_handlers.c.
>     >
>     >
>     >     Cheers,
>     >
>     >     --Timo


From henning.westerholt at 1und1.de  Mon Jun  6 12:47:34 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Mon, 6 Jun 2011 12:47:34 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
Message-ID: <201106061247.34605.henning.westerholt@1und1.de>

On Monday 06 June 2011, I?aki Baz Castillo wrote:
> [..]
> I want to match the RURI user, let's suppose $rU = "138787686223". In
> this case the above regexp matches groups 1 and 2, but since group 2
> matching is longest ("13" is longest than "1") I want to get value 2.

Hi I?aki,

if you just want to have the longest match on a prefix tree, this is what 
carrierroute was build for.
 
> In case $rU = "999999" I need to get "null", in case $rU =
> "1000022323" I need to get "1".

Its works a bit differently, but this is similar to the logic that we use. 
You've a bunch of specific prefixes with special routing, normal country 
prefixes that you want to route differently and then a default routing for the 
rest.

Best regards,

Henning


From ibc at aliax.net  Mon Jun  6 12:50:02 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 12:50:02 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <201106061247.34605.henning.westerholt@1und1.de>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
Message-ID: <BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>

2011/6/6 Henning Westerholt <henning.westerholt at 1und1.de>:
> if you just want to have the longest match on a prefix tree, this is what
> carrierroute was build for.
>
>> In case $rU = "999999" I need to get "null", in case $rU =
>> "1000022323" I need to get "1".
>
> Its works a bit differently, but this is similar to the logic that we use.
> You've a bunch of specific prefixes with special routing, normal country
> prefixes that you want to route differently and then a default routing for the
> rest

Thanks Henning. I do know that CR and LCR operate with longest prefix
matching, but I don't want this feature now for routing purposes, but
for blacklist purposes. So using dialplan module with higher
priorities in longer prefixes would be valid in my case.

Thanks a lot.

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


From henning.westerholt at 1und1.de  Mon Jun  6 13:28:17 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Mon, 6 Jun 2011 13:28:17 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
Message-ID: <201106061328.17947.henning.westerholt@1und1.de>

On Monday 06 June 2011, I?aki Baz Castillo wrote:
> > Its works a bit differently, but this is similar to the logic that we
> > use. You've a bunch of specific prefixes with special routing, normal
> > country prefixes that you want to route differently and then a default
> > routing for the rest
> 
> Thanks Henning. I do know that CR and LCR operate with longest prefix
> matching, but I don't want this feature now for routing purposes, but
> for blacklist purposes. So using dialplan module with higher
> priorities in longer prefixes would be valid in my case.

Hi I?aki,

ah, for blacklisting purposes, i see. Then dialplan should be indeed a good 
choice then. We use the userblacklist module for this, it works with the same 
longest prefix match logic as cr.

Best regards,

Henning


From ibc at aliax.net  Mon Jun  6 13:30:39 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 13:30:39 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <201106061328.17947.henning.westerholt@1und1.de>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
	<201106061328.17947.henning.westerholt@1und1.de>
Message-ID: <BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>

2011/6/6 Henning Westerholt <henning.westerholt at 1und1.de>:
> We use the userblacklist module for this, it works with the same
> longest prefix match logic as cr.

Ops, I missed that module :)
I'll take a look to it. Thanks.

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


From ibc at aliax.net  Mon Jun  6 13:34:40 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 13:34:40 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
	<201106061328.17947.henning.westerholt@1und1.de>
	<BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
Message-ID: <BANLkTi=fhYuOUoG9VmM2xDNzorg1WW3uAA@mail.gmail.com>

2011/6/6 I?aki Baz Castillo <ibc at aliax.net>:
> 2011/6/6 Henning Westerholt <henning.westerholt at 1und1.de>:
>> We use the userblacklist module for this, it works with the same
>> longest prefix match logic as cr.
>
> Ops, I missed that module :)
> I'll take a look to it. Thanks.

It's not valid for my custom case as I need a function returning an id
of the matched groups. This is, I have various geo-zone groups:

  1: Europe (prefixes of Europe)
  2: North America (prefixes of North America)
  ...

Each client has a boolean field to set permission for calls to each geo-zones.

So I think it's better for me to use dialplan module and return the
geo-zone id in the "attrs" AVP.

Thanks.

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


From henning.westerholt at 1und1.de  Mon Jun  6 13:36:02 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Mon, 6 Jun 2011 13:36:02 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTi=fhYuOUoG9VmM2xDNzorg1WW3uAA@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
	<BANLkTi=fhYuOUoG9VmM2xDNzorg1WW3uAA@mail.gmail.com>
Message-ID: <201106061336.02742.henning.westerholt@1und1.de>

On Monday 06 June 2011, I?aki Baz Castillo wrote:
> It's not valid for my custom case as I need a function returning an id
> of the matched groups. This is, I have various geo-zone groups:
> 
>   1: Europe (prefixes of Europe)
>   2: North America (prefixes of North America)
>   ...
> 
> Each client has a boolean field to set permission for calls to each
> geo-zones.
> 
> So I think it's better for me to use dialplan module and return the
> geo-zone id in the "attrs" AVP.

Yes - the dialplan is much more flexible of course. :-)

Henning


From jh at tutpro.com  Mon Jun  6 13:44:31 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Mon, 6 Jun 2011 14:44:31 +0300
Subject: [SR-Users] Using =~ to get the group identifier with
	longest	match
In-Reply-To: <BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
	<201106061328.17947.henning.westerholt@1und1.de>
	<BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
Message-ID: <19948.48543.216972.409989@sip.test.fi>

I?aki Baz Castillo writes:

> > We use the userblacklist module for this, it works with the same
> > longest prefix match logic as cr.

do both of them also support regular expressions, what inaki was asking
for?

-- juha


From ibc at aliax.net  Mon Jun  6 14:04:23 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 14:04:23 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <19948.48543.216972.409989@sip.test.fi>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
	<201106061328.17947.henning.westerholt@1und1.de>
	<BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
	<19948.48543.216972.409989@sip.test.fi>
Message-ID: <BANLkTi=+A6o1f-v5n7dH8otRb1qfjM3Xiw@mail.gmail.com>

2011/6/6 Juha Heinanen <jh at tutpro.com>:
> do both of them also support regular expressions, what inaki was asking
> for?

Dialplan module allows setting a regular expression per table entry
(by setting the column match_op to 1 (regexp).


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


From jh at tutpro.com  Mon Jun  6 13:58:35 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Mon, 6 Jun 2011 14:58:35 +0300
Subject: [SR-Users] Using =~ to get the group identifier with
	longest	match
In-Reply-To: <BANLkTi=+A6o1f-v5n7dH8otRb1qfjM3Xiw@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
	<201106061328.17947.henning.westerholt@1und1.de>
	<BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
	<19948.48543.216972.409989@sip.test.fi>
	<BANLkTi=+A6o1f-v5n7dH8otRb1qfjM3Xiw@mail.gmail.com>
Message-ID: <19948.49387.938050.166821@sip.test.fi>

I?aki Baz Castillo writes:
> 
> Dialplan module allows setting a regular expression per table entry
> (by setting the column match_op to 1 (regexp).

sure, but i was asking about cr and the other module that henning was
recommending.

-- juha


From ibc at aliax.net  Mon Jun  6 14:12:15 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 14:12:15 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <19948.49387.938050.166821@sip.test.fi>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<201106061247.34605.henning.westerholt@1und1.de>
	<BANLkTikaNBa58ryHTH9qAcPzUscuWQ9LCA@mail.gmail.com>
	<201106061328.17947.henning.westerholt@1und1.de>
	<BANLkTikbuymdGcsBuM1ThK6NnPqubd34Cg@mail.gmail.com>
	<19948.48543.216972.409989@sip.test.fi>
	<BANLkTi=+A6o1f-v5n7dH8otRb1qfjM3Xiw@mail.gmail.com>
	<19948.49387.938050.166821@sip.test.fi>
Message-ID: <BANLkTim3WbACHbn0S_Dpe+CgGOaArTX2ww@mail.gmail.com>

2011/6/6 Juha Heinanen <jh at tutpro.com>:
> sure, but i was asking about cr and the other module that henning was
> recommending.

Well, for sure cr and userblacklist implements regular expression
matching by taking the prefix from the DB and internally using
"^PREFIX" (I expect). Same as LCR.

The fact is that I don't need more regular expressions than that. This
is, I don't need to set a prefix like "(1|2[0123]{2})". :)

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


From david.villasmil.work at gmail.com  Mon Jun  6 14:37:51 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Jun 2011 14:37:51 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
Message-ID: <BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>

Hello I?aki,

I did this a long time ago using a db backend, i can't remember exactly how.

the table would be:

areacode, route
1                 route1
13               route2

I connected to mysql and ran something like:

select * from routes where '$rU' like concat(areacode,'%') order by
len(areacode) desc limit 1;


that works perfectly

hope it helps

David
On Mon, Jun 6, 2011 at 10:25 AM, I?aki Baz Castillo <ibc at aliax.net> wrote:

> Hi, I wonder how feasible is the folowing:
>
> I have a regular expresion like:
>
>  REGEXP = "^((1|20)|(13|3|4)|(5|6))"
>
> There are 3 matching groups:
>
>  1)   (1|20)
>  2)   (13|3|4)
>  3)   (5|6)
>
> I want to match the RURI user, let's suppose $rU = "138787686223". In
> this case the above regexp matches groups 1 and 2, but since group 2
> matching is longest ("13" is longest than "1") I want to get value 2.
>
> In case $rU = "999999" I need to get "null", in case $rU =
> "1000022323" I need to get "1".
>
> AFAIK there is no way to do it with current regular expression utils
> in Kamailio, but maybe I miss something. Thanks a lot.
>
>
> --
> 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
>
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From ibc at aliax.net  Mon Jun  6 14:52:13 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 14:52:13 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
Message-ID: <BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>

2011/6/6 David Villasmil <david.villasmil.work at gmail.com>:
> I connected to mysql and ran something like:
> select * from routes where '$rU' like concat(areacode,'%') order by
> len(areacode) desc limit 1;

Hi David, that would work indeed, but I prefer not to saturate the DB
with such a query which cannot use table indexes. My kamailio handles
2000-3000 concurrent calls and such calls come from callcenters so
they are very "aggressive" (maybe 200-300 calls in the same second),
so querying the database with a low performance query is not an option
in my case.

Thanks.

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


From david.villasmil.work at gmail.com  Mon Jun  6 15:05:21 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Jun 2011 15:05:21 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
Message-ID: <BANLkTikMO-e4OCiKeKZDUj_t9Ec8Pz75xA@mail.gmail.com>

Hello I?aki,

I had much the same traffic, and with a good index that wouldn't be a
problem, IMHO.
You could also load the table in memory, can't be faster than that.

Of course, doing it in the config script is faster, but it limits your
flexibility... just a thought...
I even got the rate and created a "custom" cdr for each call. we had around
4k concurrent calls at that time (like 5 years ago)

Anyway, good luck!

David

On Mon, Jun 6, 2011 at 2:52 PM, I?aki Baz Castillo <ibc at aliax.net> wrote:

> 2011/6/6 David Villasmil <david.villasmil.work at gmail.com>:
> > I connected to mysql and ran something like:
> > select * from routes where '$rU' like concat(areacode,'%') order by
> > len(areacode) desc limit 1;
>
> Hi David, that would work indeed, but I prefer not to saturate the DB
> with such a query which cannot use table indexes. My kamailio handles
> 2000-3000 concurrent calls and such calls come from callcenters so
> they are very "aggressive" (maybe 200-300 calls in the same second),
> so querying the database with a low performance query is not an option
> in my case.
>
> Thanks.
>
> --
> 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
>
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From miconda at gmail.com  Mon Jun  6 15:08:56 2011
From: miconda at gmail.com (Daniel-Constantine Mierla)
Date: Mon, 6 Jun 2011 15:08:56 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
Message-ID: <90918669-4785-45B9-BE61-5294B3963733@gmail.com>



On Jun 6, 2011, at 2:52 PM, I?aki Baz Castillo <ibc at aliax.net> wrote:

> 2011/6/6 David Villasmil <david.villasmil.work at gmail.com>:
>> I connected to mysql and ran something like:
>> select * from routes where '$rU' like concat(areacode,'%') order by
>> len(areacode) desc limit 1;
> 
> Hi David, that would work indeed, but I prefer not to saturate the DB
> with such a query which cannot use table indexes. My kamailio handles
> 2000-3000 concurrent calls and such calls come from callcenters so
> they are very "aggressive" (maybe 200-300 calls in the same second),
> so querying the database with a low performance query is not an option
> in my case.

Btw, very lightweight module for longest prefix matching is mtree. It has option to return a value in a cfg variable, that can be the group id for example. It does caching in memory trees, so it is really fast in matching.

Cheers,
Daniel

> 
> Thanks.
> 
> -- 
> 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 ibc at aliax.net  Mon Jun  6 15:20:39 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 15:20:39 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTikMO-e4OCiKeKZDUj_t9Ec8Pz75xA@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<BANLkTikMO-e4OCiKeKZDUj_t9Ec8Pz75xA@mail.gmail.com>
Message-ID: <BANLkTinDCXgeguV+2LC0xk5mGrUwJjCuRA@mail.gmail.com>

2011/6/6 David Villasmil <david.villasmil.work at gmail.com>:
> I had much the same traffic, and with a good index that wouldn't be a
> problem, IMHO.

Hi David, your query cannot make usage of table indexes:

  select * from routes where '$rU' like concat(areacode,'%') order by
len(areacode) desc limit 1;

You are using "LIKE" and concat(TABLE_COLUMN...) so you are forcing
the database server to read *all* the values of "areacode" (all the
rows), create a new string by concatening "%" at the end, and then
performing a regular expressi?n (LIKE "XXX%") for *every* resulting
values. This can never use an index.

You can check it by yourself by entering into mysql console:

  DESCRIBE select * from routes where '999999999' like
concat(areacode,'%') order by len(areacode) desc limit 1;

You will realize that no index is being used.




> You could also load the table in memory, can't be faster than that.

Yes, but for that a custom module is required :)


> Of course, doing it in the config script is faster, but it limits your
> flexibility... just a thought...

Humm, in fact not. I can manage the table content via a web interface
(or whatever) and running a MI "reload" command for the used module so
the table content is read again into memory. This is the same as I do
with LCR module and others.


Regards.



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


From ibc at aliax.net  Mon Jun  6 15:20:59 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 15:20:59 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <90918669-4785-45B9-BE61-5294B3963733@gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
Message-ID: <BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>

2011/6/6 Daniel-Constantine Mierla <miconda at gmail.com>:
> Btw, very lightweight module for longest prefix matching is mtree. It has option to return a value in a cfg variable, that can be the group id for example. It does caching in memory trees, so it is really fast in matching.

I also missed that module. Let's take a look to it.

Thanks.

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


From ibc at aliax.net  Mon Jun  6 15:29:14 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 15:29:14 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
Message-ID: <BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>

2011/6/6 I?aki Baz Castillo <ibc at aliax.net>:
> 2011/6/6 Daniel-Constantine Mierla <miconda at gmail.com>:
>> Btw, very lightweight module for longest prefix matching is mtree. It has option to return a value in a cfg variable, that can be the group id for example. It does caching in memory trees, so it is really fast in matching.
>
> I also missed that module. Let's take a look to it.

Hi Daniel, sounds really specific for which I need :)
Let me ask some doubts:

- You say "It has option to return a value in a cfg variable, that can
be the group id for example". Do you mean pv_value AVP?:
    http://kamailio.org/docs/modules/3.1.x/modules/mtree.html#id2845134
  Would that AVP be filled with the content of the "value" column for
the matched row?

- Which is the meaning of "payload" here?:
    http://kamailio.org/docs/modules/3.1.x/modules/mtree.html#id2807543

- What does mt_match() return in case there is match? and when there is not?


Thanks a lot.

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


From mino.haluz at gmail.com  Mon Jun  6 16:29:24 2011
From: mino.haluz at gmail.com (Mino Haluz)
Date: Mon, 6 Jun 2011 16:29:24 +0200
Subject: [SR-Users] Multiple INVITEs and discarded call_control
Message-ID: <BANLkTim_pCyDB8ecTkzRbT6EjniTZpit7g@mail.gmail.com>

Hi,

my kamailio server is receiving from some customers 3 identical INVITEs when
call is initiated (separated by 200ms). Those 3 INVITEs are making a big
problem with call_control:

WARNING: call_control [call_control.c:1156]: dialog to trace controlled call
was not created. discarding callcontrol.

That is why, the prepaid limit is not working at all in this case. This way
the user can hack the prepaid protection of the account. Otherwise the
call_control is fuilly functional.

Anybody experienced the similar problem? If so, how to resolve it?

Thanks,
Mino Haluz
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From miconda at gmail.com  Mon Jun  6 16:33:56 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 16:33:56 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
 match
In-Reply-To: <BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
Message-ID: <4DECE554.5010407@gmail.com>

Hello,

On 6/6/11 3:29 PM, I?aki Baz Castillo wrote:
> 2011/6/6 I?aki Baz Castillo<ibc at aliax.net>:
>> 2011/6/6 Daniel-Constantine Mierla<miconda at gmail.com>:
>>> Btw, very lightweight module for longest prefix matching is mtree. It has option to return a value in a cfg variable, that can be the group id for example. It does caching in memory trees, so it is really fast in matching.
>> I also missed that module. Let's take a look to it.
> Hi Daniel, sounds really specific for which I need :)
> Let me ask some doubts:
>
> - You say "It has option to return a value in a cfg variable, that can
> be the group id for example". Do you mean pv_value AVP?:
>      http://kamailio.org/docs/modules/3.1.x/modules/mtree.html#id2845134
yes, but it can be any kind of writable variable, not restricted to AVP 
-- see the example in the readme where it is a $var()
>    Would that AVP be filled with the content of the "value" column for
> the matched row?
Exactly
>
> - Which is the meaning of "payload" here?:
>      http://kamailio.org/docs/modules/3.1.x/modules/mtree.html#id2807543
when it is 0, will return the value as string in varibale. The goal is 
to parse in cache the value into some internal structure. Right now 
there is also the option of 1, expecting some 'id=value', both being 
integers, with the purpose of returning all matches for the prefix 
ordered by value. It hasn't be tested extensively and thus not 
documented - initially work was on hold to get first xavps so the 
structure can be returned nicely to config, then lack of time kept it 
posponed.
>
> - What does mt_match() return in case there is match? and when there is not?
True and false.

Cheers,
Daniel

>
>
> Thanks a lot.
>

-- 
Daniel-Constantin Mierla
http://www.asipto.com



From ibc at aliax.net  Mon Jun  6 17:13:29 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 17:13:29 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <4DECE554.5010407@gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
	<4DECE554.5010407@gmail.com>
Message-ID: <BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>

2011/6/6 Daniel-Constantin Mierla <miconda at gmail.com>:
>> - You say "It has option to return a value in a cfg variable, that can
>> be the group id for example". Do you mean pv_value AVP?:
>> ? ? http://kamailio.org/docs/modules/3.1.x/modules/mtree.html#id2845134
>
> yes, but it can be any kind of writable variable, not restricted to AVP --
> see the example in the readme where it is a $var()
>>
>> ? Would that AVP be filled with the content of the "value" column for
>> the matched row?
>
> Exactly
>>
>> - Which is the meaning of "payload" here?:
>> ? ? http://kamailio.org/docs/modules/3.1.x/modules/mtree.html#id2807543
>
> when it is 0, will return the value as string in varibale. The goal is to
> parse in cache the value into some internal structure. Right now there is
> also the option of 1, expecting some 'id=value', both being integers, with
> the purpose of returning all matches for the prefix ordered by value. It
> hasn't be tested extensively and thus not documented - initially work was on
> hold to get first xavps so the structure can be returned nicely to config,
> then lack of time kept it posponed.
>>
>> - What does mt_match() return in case there is match? and when there is
>> not?
>
> True and false.


Thanks Daniel, please let me one question more: does the mtree module
allow setting two different tables (with maybe also different columns
names)? The function mt_match(mtree, pv, mode) seems to allow, as
first argument, the "model/table" to use, but I see no way of defining
different models, is it?

Thanks a lot.

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


From ibc at aliax.net  Mon Jun  6 17:23:28 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 17:23:28 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
	<4DECE554.5010407@gmail.com>
	<BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>
Message-ID: <BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>

2011/6/6 I?aki Baz Castillo <ibc at aliax.net>:
>
> Thanks Daniel, please let me one question more: does the mtree module
> allow setting two different tables (with maybe also different columns
> names)? The function mt_match(mtree, pv, mode) seems to allow, as
> first argument, the "model/table" to use, but I see no way of defining
> different models, is it?

Ok, it seems that using "mtree" parameter does the job (but columns
must have same name in all the tables):

  modparam("mtree", "mtree", "name=mytable;dbtable=routes;type=0;")

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


From ibc at aliax.net  Mon Jun  6 17:26:59 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 17:26:59 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
	<4DECE554.5010407@gmail.com>
	<BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>
	<BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>
Message-ID: <BANLkTinbczw5sQWCVTDHLJpXijc0UjULbw@mail.gmail.com>

2011/6/6 I?aki Baz Castillo <ibc at aliax.net>:
>> Thanks Daniel, please let me one question more: does the mtree module
>> allow setting two different tables (with maybe also different columns
>> names)? The function mt_match(mtree, pv, mode) seems to allow, as
>> first argument, the "model/table" to use, but I see no way of defining
>> different models, is it?
>
> Ok, it seems that using "mtree" parameter does the job (but columns
> must have same name in all the tables):
>
> ?modparam("mtree", "mtree", "name=mytable;dbtable=routes;type=0;")

Sorry for so many questions but, what is "tname_column"?:

  3.4. tname_column (string)
  Name of 'tname' column.

The table scheme clearly shows id, tprefix and tvalue columns,
tprefix_column and tvalue_column parameters can be used, as usual, to
rename such columns but, what is "tname_column"?

Thanks.

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


From miconda at gmail.com  Mon Jun  6 17:27:54 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 17:27:54 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
 match
In-Reply-To: <BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
	<4DECE554.5010407@gmail.com>
	<BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>
	<BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>
Message-ID: <4DECF1FA.1090207@gmail.com>



On 6/6/11 5:23 PM, I?aki Baz Castillo wrote:
> 2011/6/6 I?aki Baz Castillo<ibc at aliax.net>:
>> Thanks Daniel, please let me one question more: does the mtree module
>> allow setting two different tables (with maybe also different columns
>> names)? The function mt_match(mtree, pv, mode) seems to allow, as
>> first argument, the "model/table" to use, but I see no way of defining
>> different models, is it?
> Ok, it seems that using "mtree" parameter does the job (but columns
> must have same name in all the tables):
>
>    modparam("mtree", "mtree", "name=mytable;dbtable=routes;type=0;")
yes columns names are configurable per module -- maybe in the future 
will be per mtree.

Meanwhile maybe you can use mysql views to get to the column names 
expected by mtree.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Mon Jun  6 17:29:38 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 17:29:38 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
 match
In-Reply-To: <BANLkTinbczw5sQWCVTDHLJpXijc0UjULbw@mail.gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
	<4DECE554.5010407@gmail.com>
	<BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>
	<BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>
	<BANLkTinbczw5sQWCVTDHLJpXijc0UjULbw@mail.gmail.com>
Message-ID: <4DECF262.6000906@gmail.com>



On 6/6/11 5:26 PM, I?aki Baz Castillo wrote:
> 2011/6/6 I?aki Baz Castillo<ibc at aliax.net>:
>>> Thanks Daniel, please let me one question more: does the mtree module
>>> allow setting two different tables (with maybe also different columns
>>> names)? The function mt_match(mtree, pv, mode) seems to allow, as
>>> first argument, the "model/table" to use, but I see no way of defining
>>> different models, is it?
>> Ok, it seems that using "mtree" parameter does the job (but columns
>> must have same name in all the tables):
>>
>>   modparam("mtree", "mtree", "name=mytable;dbtable=routes;type=0;")
> Sorry for so many questions but, what is "tname_column"?:
>
>    3.4. tname_column (string)
>    Name of 'tname' column.
>
> The table scheme clearly shows id, tprefix and tvalue columns,
> tprefix_column and tvalue_column parameters can be used, as usual, to
> rename such columns but, what is "tname_column"?
The module can load on db table in a memory tree or one db table in many 
trees. For the second case, the db table name is a module parameter and 
in addition it must have an extra column to specify the memory tree name.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Mon Jun  6 17:37:13 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 6 Jun 2011 17:37:13 +0200
Subject: [SR-Users] Using =~ to get the group identifier with longest
	match
In-Reply-To: <4DECF262.6000906@gmail.com>
References: <BANLkTik0DzvJtRpphi+tZXA0Y9B==t9XYA@mail.gmail.com>
	<BANLkTinN0V7H9=a5kbh4x-z3gQj8kARjmw@mail.gmail.com>
	<BANLkTin=8=88m9y-0BgrhYVrqSWJKrDLHQ@mail.gmail.com>
	<90918669-4785-45B9-BE61-5294B3963733@gmail.com>
	<BANLkTinfT3-G-ULViO4Dbc8pcnD28e5AUQ@mail.gmail.com>
	<BANLkTi=uq4ubx3ZUWL00aCo=V-O_8CCUrw@mail.gmail.com>
	<4DECE554.5010407@gmail.com>
	<BANLkTikS2r03urciUbx0Y5AFL0Fw287CnQ@mail.gmail.com>
	<BANLkTikS=mCEG4KCX8Fizu2M3xDJh+K_0w@mail.gmail.com>
	<BANLkTinbczw5sQWCVTDHLJpXijc0UjULbw@mail.gmail.com>
	<4DECF262.6000906@gmail.com>
Message-ID: <BANLkTindKwix5PP=UY+Sr4Y4gidOzFdLaQ@mail.gmail.com>

2011/6/6 Daniel-Constantin Mierla <miconda at gmail.com>:
> The module can load on db table in a memory tree or one db table in many
> trees. For the second case, the db table name is a module parameter and in
> addition it must have an extra column to specify the memory tree name.

Great. So we can extract different mtrees from the same table, just by
filtering by tname column (and then using the assigned name for the
module function). Just great :)

Thanks a lot.

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


From skchopperguy at gmail.com  Mon Jun  6 20:27:29 2011
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 6 Jun 2011 11:27:29 -0700
Subject: [SR-Users] Error bad syntax - DEBUG_FLAGS
Message-ID: <01e701cc2477$6575f840$3061e8c0$@com>

Hi all,

 

 I'm new to Kamailio (3.1), learning the configuration language. I'm trying
to take the advice from one of the docs to create a DEBUG block and place
calls everywhere in my script. Right out of the gate I get these errors on
loading.  

 

 Can someone tell me where I went wrong and how I can fix this? It says flag
not declared though it's there and I'm a bit lost.

 

 

0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 6: syntax error

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' or ')' ?

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 6: bad command: missing ';'?

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 18: '('')' expected (function
call)

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 18: bad command: missing ';'?

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 20: bad command

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 802, column 24: flag not declared

 

#------------------------------

# Flags

#------------------------------

#!define FLAG_DEBUG 1

...

route {

 

               # Call debug flags

                route(ROUTE_DEBUG_FLAGS);

 

.....

 

route[ROUTE_DEBUG_FLAGS] {

  resetflag(FLAG_DEBUG);

  #------------------------------------------------------------

  # Methods that will have debugging turned on

  #-----------------------------------------------------------

       if (is_method("ACK")) { resetflag(FLAG_DEBUG); }

 else if (is_method("BYE")) { setflag(FLAG_DEBUG); }

 else if (is_method("CANCEL")) { setflag(FLAG_DEBUG); }

 else if (is_method("INVITE"))        { setflag(FLAG_DEBUG); }

  else if (is_method("OPTIONS"))       { setflag(FLAG_DEBUG); }

  else if (is_method("PUBLISH"))       { setflag(FLAG_DEBUG); }

  else if (is_method("REGISTER"))      { resetflag(FLAG_DEBUG); }

  else if (is_method("MESSAGE"))       { setflag(FLAG_DEBUG); }

  else if (is_method("NOTIFY"))        { setflag(FLAG_DEBUG); }

  else if (is_method("SUBSCRIBE"))     { setflag(FLAG_DEBUG); }

  else if (is_method("INFO"))          { setflag(FLAG_DEBUG); }

  else if (is_method("PRACK"))         { setflag(FLAG_DEBUG); }

  else if (is_method("REFER"))         { setflag(FLAG_DEBUG); }

  else {

     setflag(FLAG_DEBUG);

  }

}

 

}

 

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From miconda at gmail.com  Mon Jun  6 22:13:34 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 22:13:34 +0200
Subject: [SR-Users] Error bad syntax - DEBUG_FLAGS
In-Reply-To: <01e701cc2477$6575f840$3061e8c0$@com>
References: <01e701cc2477$6575f840$3061e8c0$@com>
Message-ID: <4DED34EE.3060802@gmail.com>

Hello,

if you don't attach the configuration file, then you have to tell which 
are the lines 800 and 802.

 From what you pasted, seems to be at least an extra curly brace at the 
end of route[ROUTE_DEBUG_FLAGS].

Cheers,
Daniel

On 6/6/11 8:27 PM, Skyler wrote:
>
> Hi all,
>
>  I'm new to Kamailio (3.1), learning the configuration language. I'm 
> trying to take the advice from one of the docs to create a DEBUG block 
> and place calls everywhere in my script. Right out of the gate I get 
> these errors on loading.
>
>  Can someone tell me where I went wrong and how I can fix this? It 
> says flag not declared though it's there and I'm a bit lost.
>
> 0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 800, column 6: syntax error
>
>  0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' or ')' ?
>
>  0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 800, column 6: bad command: missing ';'?
>
>  0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 800, column 18: '('')' expected 
> (function call)
>
>  0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 800, column 18: bad command: missing 
> ';'?
>
>  0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 800, column 20: bad command
>
>  0(10986) : <core> [cfg.y:3412]: parse error in config file 
> //etc/kamailio/kamailio.cfg, line 802, column 24: flag not declared
>
> #------------------------------
>
> # Flags
>
> #------------------------------
>
> #!define FLAG_DEBUG 1
>
> .......
>
> route {
>
>                # Call debug flags
>
>                 route(ROUTE_DEBUG_FLAGS);
>
> ...............
>
> route[ROUTE_DEBUG_FLAGS] {
>
>   resetflag(FLAG_DEBUG);
>
>   #------------------------------------------------------------
>
>   # Methods that will have debugging turned on
>
>   #-----------------------------------------------------------
>
>        if (is_method("ACK")) { resetflag(FLAG_DEBUG); }
>
>  else if (is_method("BYE")) { setflag(FLAG_DEBUG); }
>
>  else if (is_method("CANCEL")) { setflag(FLAG_DEBUG); }
>
>  else if (is_method("INVITE"))        { setflag(FLAG_DEBUG); }
>
>   else if (is_method("OPTIONS"))       { setflag(FLAG_DEBUG); }
>
>   else if (is_method("PUBLISH"))       { setflag(FLAG_DEBUG); }
>
>   else if (is_method("REGISTER"))      { resetflag(FLAG_DEBUG); }
>
>   else if (is_method("MESSAGE"))       { setflag(FLAG_DEBUG); }
>
>   else if (is_method("NOTIFY"))        { setflag(FLAG_DEBUG); }
>
>   else if (is_method("SUBSCRIBE"))     { setflag(FLAG_DEBUG); }
>
>   else if (is_method("INFO"))          { setflag(FLAG_DEBUG); }
>
>   else if (is_method("PRACK"))         { setflag(FLAG_DEBUG); }
>
>   else if (is_method("REFER"))         { setflag(FLAG_DEBUG); }
>
>   else {
>
>      setflag(FLAG_DEBUG);
>
>   }
>
> }
>
> }
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From miconda at gmail.com  Mon Jun  6 22:16:58 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 22:16:58 +0200
Subject: [SR-Users] Multiple INVITEs and discarded call_control
In-Reply-To: <BANLkTim_pCyDB8ecTkzRbT6EjniTZpit7g@mail.gmail.com>
References: <BANLkTim_pCyDB8ecTkzRbT6EjniTZpit7g@mail.gmail.com>
Message-ID: <4DED35BA.8020901@gmail.com>

Hello,

if it is a retransmission, then just do t_newtran() in your config file 
before handling it to call control to be sure the retransmitted requests 
are absorbed.
http://kamailio.org/docs/modules/stable/modules/tm.html#t_newtran

Cheers,
Daniel

On 6/6/11 4:29 PM, Mino Haluz wrote:
> Hi,
>
> my kamailio server is receiving from some customers 3 identical 
> INVITEs when call is initiated (separated by 200ms). Those 3 INVITEs 
> are making a big problem with call_control:
>
> WARNING: call_control [call_control.c:1156]: dialog to trace 
> controlled call was not created. discarding callcontrol.
>
> That is why, the prepaid limit is not working at all in this case. 
> This way the user can hack the prepaid protection of the account. 
> Otherwise the call_control is fuilly functional.
>
> Anybody experienced the similar problem? If so, how to resolve it?
>
> Thanks,
> Mino Haluz
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From skchopperguy at gmail.com  Mon Jun  6 23:05:15 2011
From: skchopperguy at gmail.com (Skyler)
Date: Mon, 6 Jun 2011 14:05:15 -0700
Subject: [SR-Users] Error bad syntax - DEBUG_FLAGS
In-Reply-To: <4DED34EE.3060802@gmail.com>
References: <01e701cc2477$6575f840$3061e8c0$@com> <4DED34EE.3060802@gmail.com>
Message-ID: <021f01cc248d$6fdc29b0$4f947d10$@com>

Hello Daniel,

 

 Thanks for the reply. The curly brace was to close the route {. I have
since commented out the block entirely and found many more error messages,
so this is not actually the source of my trouble. I obviously skipped ahead
without testing something first and now it is hiding a problem. 

 Reverting to my backup now and starting over.

 

S.

 

From: Daniel-Constantin Mierla [mailto:miconda at gmail.com] 
Sent: Monday, June 06, 2011 1:14 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Cc: Skyler
Subject: Re: [SR-Users] Error bad syntax - DEBUG_FLAGS

 

Hello,

if you don't attach the configuration file, then you have to tell which are
the lines 800 and 802.


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.0.1307394330.2669.sr-users@lists.sip-router.org>

of route[ROUTE_DEBUG_FLAGS].

Cheers,
Daniel

On 6/6/11 8:27 PM, Skyler wrote: 

Hi all,

 

 I'm new to Kamailio (3.1), learning the configuration language. I'm trying
to take the advice from one of the docs to create a DEBUG block and place
calls everywhere in my script. Right out of the gate I get these errors on
loading.  

 

 Can someone tell me where I went wrong and how I can fix this? It says flag
not declared though it's there and I'm a bit lost.

 

 

0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 6: syntax error

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' or ')' ?

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 6: bad command: missing ';'?

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 18: '('')' expected (function
call)

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 18: bad command: missing ';'?

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 800, column 20: bad command

 0(10986) : <core> [cfg.y:3412]: parse error in config file
//etc/kamailio/kamailio.cfg, line 802, column 24: flag not declared

 

#------------------------------

# Flags

#------------------------------

#!define FLAG_DEBUG 1

...

route {

 

               # Call debug flags

                route(ROUTE_DEBUG_FLAGS);

 

.....

 

route[ROUTE_DEBUG_FLAGS] {

  resetflag(FLAG_DEBUG);

  #------------------------------------------------------------

  # Methods that will have debugging turned on

  #-----------------------------------------------------------

       if (is_method("ACK")) { resetflag(FLAG_DEBUG); }

 else if (is_method("BYE")) { setflag(FLAG_DEBUG); }

 else if (is_method("CANCEL")) { setflag(FLAG_DEBUG); }

 else if (is_method("INVITE"))        { setflag(FLAG_DEBUG); }

  else if (is_method("OPTIONS"))       { setflag(FLAG_DEBUG); }

  else if (is_method("PUBLISH"))       { setflag(FLAG_DEBUG); }

  else if (is_method("REGISTER"))      { resetflag(FLAG_DEBUG); }

  else if (is_method("MESSAGE"))       { setflag(FLAG_DEBUG); }

  else if (is_method("NOTIFY"))        { setflag(FLAG_DEBUG); }

  else if (is_method("SUBSCRIBE"))     { setflag(FLAG_DEBUG); }

  else if (is_method("INFO"))          { setflag(FLAG_DEBUG); }

  else if (is_method("PRACK"))         { setflag(FLAG_DEBUG); }

  else if (is_method("REFER"))         { setflag(FLAG_DEBUG); }

  else {

     setflag(FLAG_DEBUG);

  }

}

 

}

 






_______________________________________________
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://linkedin.com/in/miconda -- http://twitter.com/miconda

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<p class=3DMsoNormal><span style=3D'color:#1F497D'>Hello =
Daniel,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;Thanks for the =
reply. The curly
brace was to close the route {. I have since commented out the block =
entirely and
found many more error messages, so this is not actually the source of my
trouble. I obviously skipped ahead without testing something first and =
now it
is hiding a problem. <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;Reverting to my =
backup now
and starting over.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'>S.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> Daniel-Constantin Mierla
[mailto:miconda at gmail.com] <br>
<b>Sent:</b> Monday, June 06, 2011 1:14 PM<br>
<b>To:</b> SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) =
- Users
Mailing List<br>
<b>Cc:</b> Skyler<br>
<b>Subject:</b> Re: [SR-Users] Error bad syntax - =
DEBUG_FLAGS<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hello,<br>
<br>
if you don't attach the configuration file, then you have to tell which =
are the
lines 800 and 802.<br>
<br>

From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.1.1307394330.2669.sr-users@lists.sip-router.org>

end of
route[ROUTE_DEBUG_FLAGS].<br>
<br>
Cheers,<br>
Daniel<br>
<br>
On 6/6/11 8:27 PM, Skyler wrote: <o:p></o:p></p>

<p class=3DMsoNormal>Hi all,<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;I&#8217;m new to Kamailio (3.1), learning the
configuration language. I&#8217;m trying to take the advice from one of =
the
docs to create a DEBUG block and place calls everywhere in my script. =
Right out
of the gate I get these errors on loading. &nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;Can someone tell me where I went wrong and =
how I can
fix this? It says flag not declared though it&#8217;s there and =
I&#8217;m a bit
lost.<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>0(10986) : &lt;core&gt; [cfg.y:3412]: parse error =
in config
file //etc/kamailio/kamailio.cfg, line 800, column 6: syntax =
error<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;0(10986) : &lt;core&gt; [cfg.y:3412]: parse =
error in
config file //etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' =
or ')'
?<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;0(10986) : &lt;core&gt; [cfg.y:3412]: parse =
error in
config file //etc/kamailio/kamailio.cfg, line 800, column 6: bad =
command:
missing ';'?<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;0(10986) : &lt;core&gt; [cfg.y:3412]: parse =
error in
config file //etc/kamailio/kamailio.cfg, line 800, column 18: '('')' =
expected
(function call)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;0(10986) : &lt;core&gt; [cfg.y:3412]: parse =
error in
config file //etc/kamailio/kamailio.cfg, line 800, column 18: bad =
command:
missing ';'?<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;0(10986) : &lt;core&gt; [cfg.y:3412]: parse =
error in
config file //etc/kamailio/kamailio.cfg, line 800, column 20: bad =
command<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;0(10986) : &lt;core&gt; [cfg.y:3412]: parse =
error in
config file //etc/kamailio/kamailio.cfg, line 802, column 24: flag not =
declared<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>#------------------------------<o:p></o:p></p>

<p class=3DMsoNormal># Flags<o:p></o:p></p>

<p class=3DMsoNormal>#------------------------------<o:p></o:p></p>

<p class=3DMsoNormal>#!define FLAG_DEBUG 1<o:p></o:p></p>

<p class=3DMsoNormal>&#8230;&#8230;.<o:p></o:p></p>

<p class=3DMsoNormal>route {<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p =
class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
# Call debug flags<o:p></o:p></p>

<p =
class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
route(ROUTE_DEBUG_FLAGS);<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&#8230;&#8230;&#8230;&#8230;&#8230;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>route[ROUTE_DEBUG_FLAGS] {<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; resetflag(FLAG_DEBUG);<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;
#------------------------------------------------------------<o:p></o:p><=
/p>

<p class=3DMsoNormal>&nbsp; # Methods that will have debugging turned =
on<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;
#-----------------------------------------------------------<o:p></o:p></=
p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; if
(is_method(&quot;ACK&quot;)) { resetflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;else if (is_method(&quot;BYE&quot;)) {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;else if (is_method(&quot;CANCEL&quot;)) {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;else if
(is_method(&quot;INVITE&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;OPTIONS&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;PUBLISH&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;REGISTER&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; {
resetflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;MESSAGE&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;NOTIFY&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;SUBSCRIBE&quot;))&nbsp;&nbsp;&nbsp;&nbsp; {
setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;INFO&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;
{ setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;PRACK&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
{ setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else if
(is_method(&quot;REFER&quot;))&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
{ setflag(FLAG_DEBUG); }<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; else {<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp; =
setflag(FLAG_DEBUG);<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp; }<o:p></o:p></p>

<p class=3DMsoNormal>}<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>}<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'><br>
<br>
<br>
<o:p></o:p></span></p>

<pre>_______________________________________________<o:p></o:p></pre><pre=
>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing =
list<o:p></o:p></pre><pre><a
href=3D"mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.o=
rg</a><o:p></o:p></pre><pre><a
href=3D"http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">ht=
tp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p=
></pre>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times =
New Roman","serif"'><br>
<br>
<o:p></o:p></span></p>

<pre>-- <o:p></o:p></pre><pre>Daniel-Constantin Mierla -- <a
href=3D"http://www.asipto.com">http://www.asipto.com</a><o:p></o:p></pre>=
<pre><a
href=3D"http://linkedin.com/in/miconda">http://linkedin.com/in/miconda</a=
> -- <a
href=3D"http://twitter.com/miconda">http://twitter.com/miconda</a><o:p></=
o:p></pre></div>

</body>

</html>

------=_NextPart_000_0220_01CC2452.C37D51B0--



From agranig at sipwise.com  Mon Jun  6 23:14:46 2011
From: agranig at sipwise.com (Andreas Granig)
Date: Mon, 06 Jun 2011 23:14:46 +0200
Subject: [SR-Users] xhttp and dollar sign
Message-ID: <4DED4346.4000801@sipwise.com>

Hi,

Is it possible to somehow quote the $ sign in xhttp_send_reply() in
order not to be interpreted by kamailio-3.1.3? Tried with "\" and
putting the whole body in single quotes, but I always get this error
message during startup:

ERROR: <core> [pvapi.c:720]: wrong char [$/36] in [${foo}]
ERROR: <core> [sr_module.c:1156]: bad PVE format: "${foo}"
ERROR: <core> [mod_fix.c:501]: Cannot convert function parameter 1 to spve

So I actually just want to return the string "${foo}".

Andreas

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From miconda at gmail.com  Mon Jun  6 23:16:43 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 06 Jun 2011 23:16:43 +0200
Subject: [SR-Users] xhttp and dollar sign
In-Reply-To: <4DED4346.4000801@sipwise.com>
References: <4DED4346.4000801@sipwise.com>
Message-ID: <4DED43BB.4040600@gmail.com>

Hello,

try this one:
http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:3.1.x#pseudo-variable_marker

Cheers,
Daniel

On 6/6/11 11:14 PM, Andreas Granig wrote:
> Hi,
>
> Is it possible to somehow quote the $ sign in xhttp_send_reply() in
> order not to be interpreted by kamailio-3.1.3? Tried with "\" and
> putting the whole body in single quotes, but I always get this error
> message during startup:
>
> ERROR:<core>  [pvapi.c:720]: wrong char [$/36] in [${foo}]
> ERROR:<core>  [sr_module.c:1156]: bad PVE format: "${foo}"
> ERROR:<core>  [mod_fix.c:501]: Cannot convert function parameter 1 to spve
>
> So I actually just want to return the string "${foo}".
>
> 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

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From agranig at sipwise.com  Tue Jun  7 00:30:11 2011
From: agranig at sipwise.com (Andreas Granig)
Date: Tue, 07 Jun 2011 00:30:11 +0200
Subject: [SR-Users] xhttp and dollar sign
In-Reply-To: <4DED43BB.4040600@gmail.com>
References: <4DED4346.4000801@sipwise.com> <4DED43BB.4040600@gmail.com>
Message-ID: <4DED54F3.5020301@sipwise.com>

Hi Daniel,

On 06/06/2011 11:16 PM, Daniel-Constantin Mierla wrote:
> try this one:
> http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:3.1.x#pseudo-variable_marker


Thanks alot!

Andreas

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From sanjeevsingh15 at gmail.com  Tue Jun  7 05:48:30 2011
From: sanjeevsingh15 at gmail.com (Sanjeev Singh)
Date: Tue, 7 Jun 2011 09:18:30 +0530
Subject: [SR-Users] t.38 fax
Message-ID: <BANLkTim4VHb4XQdHprkUCiiw2Zmta3ZUVg@mail.gmail.com>

does SER support t.38 fax?
what other features related to fax does SER support. please expain in
detail.

thanks,
Sanjeev Singh
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From mokhtar.bengana at gmail.com  Tue Jun  7 05:48:55 2011
From: mokhtar.bengana at gmail.com (Mokhtar Bengana)
Date: Mon, 6 Jun 2011 22:48:55 -0500
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
In-Reply-To: <4DECADE0.3000708@portaone.com>
References: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>
	<4DE8AA75.902@evaristesys.com>
	<BANLkTinv88kitQE3KmD9+F4ZRiGh+XAysA@mail.gmail.com>
	<4DECADE0.3000708@portaone.com>
Message-ID: <BANLkTik97n9+7ZFyYe8wx1SAvB_shOXO_Q@mail.gmail.com>

Andrew,

I am trying to use rtpproxy_offer/answer on the ONREPLY ROUTE but I am
having some syntax issues. Here is the example I am using. Any help
with the syntax appreciated.


route {
...
    if (is_method("INVITE")) {
        if (has_sdp()) {
            if (rtpproxy_offer())
                t_on_reply("1");
        } else {
            t_on_reply("2");
        }
    }
    if (is_method("ACK") && has_sdp())
        rtpproxy_answer();
...
}

onreply_route[1]
{
...
    if (has_sdp())
        rtpproxy_answer();
...
}

onreply_route[2]
{
...
    if (has_sdp())
        rtpproxy_offer();
...
}

On Mon, Jun 6, 2011 at 5:37 AM, Andrew Pogrebennyk
<andrew.pogrebennyk at portaone.com> wrote:
> Mokhtar, could you please make sure that you are calling route(RTPPROXY) in
> reply route as well, as Alex suggested?
>
> On 03.06.2011 16:33, Mokhtar Bengana wrote:
>>
>> This is how I configured rtpproxy. Not sure why rtpproxy is not
>> engaged both ways. Thanks for your help.
>
> --
> Sincerely,
> Andrew Pogrebennyk
>
> _______________________________________________
> 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 abalashov at evaristesys.com  Tue Jun  7 05:52:53 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 6 Jun 2011 23:52:53 -0400
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
In-Reply-To: <BANLkTik97n9+7ZFyYe8wx1SAvB_shOXO_Q@mail.gmail.com>
References: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>
	<4DE8AA75.902@evaristesys.com>
	<BANLkTinv88kitQE3KmD9+F4ZRiGh+XAysA@mail.gmail.com>
	<4DECADE0.3000708@portaone.com>
	<BANLkTik97n9+7ZFyYe8wx1SAvB_shOXO_Q@mail.gmail.com>
Message-ID: <8D774EE2-EDCB-4B61-886E-26ACF1E3AD63@evaristesys.com>

What are the syntax issues?

P.S. has_sdp() doesn't actually exist.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jun 6, 2011, at 11:48 PM, Mokhtar Bengana <mokhtar.bengana at gmail.com> wrote:

> Andrew,
> 
> I am trying to use rtpproxy_offer/answer on the ONREPLY ROUTE but I am
> having some syntax issues. Here is the example I am using. Any help
> with the syntax appreciated.
> 
> 
> route {
> ...
>    if (is_method("INVITE")) {
>        if (has_sdp()) {
>            if (rtpproxy_offer())
>                t_on_reply("1");
>        } else {
>            t_on_reply("2");
>        }
>    }
>    if (is_method("ACK") && has_sdp())
>        rtpproxy_answer();
> ...
> }
> 
> onreply_route[1]
> {
> ...
>    if (has_sdp())
>        rtpproxy_answer();
> ...
> }
> 
> onreply_route[2]
> {
> ...
>    if (has_sdp())
>        rtpproxy_offer();
> ...
> }
> 
> On Mon, Jun 6, 2011 at 5:37 AM, Andrew Pogrebennyk
> <andrew.pogrebennyk at portaone.com> wrote:
>> Mokhtar, could you please make sure that you are calling route(RTPPROXY) in
>> reply route as well, as Alex suggested?
>> 
>> On 03.06.2011 16:33, Mokhtar Bengana wrote:
>>> 
>>> This is how I configured rtpproxy. Not sure why rtpproxy is not
>>> engaged both ways. Thanks for your help.
>> 
>> --
>> Sincerely,
>> Andrew Pogrebennyk
>> 
>> _______________________________________________
>> 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 abalashov at evaristesys.com  Tue Jun  7 05:53:45 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 6 Jun 2011 23:53:45 -0400
Subject: [SR-Users] t.38 fax
In-Reply-To: <BANLkTim4VHb4XQdHprkUCiiw2Zmta3ZUVg@mail.gmail.com>
References: <BANLkTim4VHb4XQdHprkUCiiw2Zmta3ZUVg@mail.gmail.com>
Message-ID: <ED245636-ECD5-4DF7-9F0F-55321B54ED29@evaristesys.com>

It's a SIP proxy; it doesn't care about the media/codec and is not involved in it.  So, it "supports" everything, including T.38.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jun 6, 2011, at 11:48 PM, Sanjeev Singh <sanjeevsingh15 at gmail.com> wrote:

> does SER support t.38 fax?
> what other features related to fax does SER support. please expain in detail.
> 
> thanks,
> Sanjeev Singh 
> _______________________________________________
> 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 mokhtar.bengana at gmail.com  Wed Jun  8 04:31:17 2011
From: mokhtar.bengana at gmail.com (Mokhtar Bengana)
Date: Tue, 7 Jun 2011 21:31:17 -0500
Subject: [SR-Users] incorrect port 0 in reply from rtp proxy
In-Reply-To: <8D774EE2-EDCB-4B61-886E-26ACF1E3AD63@evaristesys.com>
References: <BANLkTimwQ4VNtyZ3zqbRM-tizXG-dQJr_w@mail.gmail.com>
	<4DE8AA75.902@evaristesys.com>
	<BANLkTinv88kitQE3KmD9+F4ZRiGh+XAysA@mail.gmail.com>
	<4DECADE0.3000708@portaone.com>
	<BANLkTik97n9+7ZFyYe8wx1SAvB_shOXO_Q@mail.gmail.com>
	<8D774EE2-EDCB-4B61-886E-26ACF1E3AD63@evaristesys.com>
Message-ID: <BANLkTinDz2rcx+U6g2GzPkM0CXk7Sm4ikw@mail.gmail.com>

I am still unable to fix the one way audio for my outgoing calls. I
tried different flags for nat_uac_test and force_rtp_proxy and nothing
made a difference.  Wireshack is not showing an rtp stream being
initiated from the callee back to my natted caller ip. I have included
my routing logic and appreciate any help figuring out why rtpproxy is
not engaged in both legs of the rtp stream.


# - flags
#   FLT_ - per transaction (message) flags
#       FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7


#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222")

# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger at mydomain.com")

# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endi


####### Routing Logic ########


# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {

        # per request initial checks
        route(REQINIT);

        # NAT detection
        route(NAT);

        # handle requests within SIP dialogs
        route(WITHINDLG);

        ### only initial requests (no To tag)

        # CANCEL processing
        if (is_method("CANCEL"))
        {
                if (t_check_trans())
                        t_relay();
                exit;
        }

        t_check_trans();

        # authentication
        route(AUTH);

        # record routing for dialog forming requests (in case they are routed)
        # - remove preloaded route headers
        remove_hf("Route");
        if (is_method("INVITE|SUBSCRIBE"))
                record_route();

        # account only INVITEs
        if (is_method("INVITE"))
        {
                setflag(FLT_ACC); # do accounting
        }

        # dispatch requests to foreign domains
        route(SIPOUT);

        ### requests for my local domains

        # handle presence related requests
        route(PRESENCE);

        # handle registrations
        route(REGISTRAR);

        if ($rU==$null)
        {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }

        # dispatch destinations to PSTN
        route(PSTN);

        # user location service
        route(LOCATION);

        route(RELAY);
}


route[RELAY] {
#!ifdef WITH_NAT
        if (check_route_param("nat=yes")) {
                setbflag(FLB_NATB);
        }
        if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
                route(RTPPROXY);
        }
#!endif

        /* example how to enable some additional event routes */
        if (is_method("INVITE")) {
                #t_on_branch("BRANCH_ONE");
                t_on_reply("REPLY_ONE");
                t_on_failure("FAIL_ONE");
        }

        if (!t_relay()) {
                sl_reply_error();
        }
        exit;
}

# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
        # flood dection from same IP and traffic ban for a while
        # be sure you exclude checking trusted peers, such as pstn gateways
        # - local host excluded (e.g., loop to self)
        if(src_ip!=myself)
        {
                if($sht(ipban=>$si)!=$null)
                {
                        # ip is already blocked
                        xdbg("request from blocked IP - $rm from $fu
(IP:$si:$sp)\n");
                        exit;
                }
                if (!pike_check_req())
                {
                        xlog("L_ALERT","ALERT: pike blocking $rm from
$fu (IP:$si:$sp)\n");
                        $sht(ipban=>$si) = 1;
                        exit;
                }
        }
#!endif

        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }

        if(!sanity_check("1511", "7"))
        {
                xlog("Malformed SIP message from $si:$sp\n");
                exit;
        }
}

# Handle requests within SIP dialogs
route[WITHINDLG] {
        if (has_totag()) {
                # sequential request withing a dialog should
                # take the path determined by record-routing
                if (loose_route()) {
                        if (is_method("BYE")) {
                                setflag(FLT_ACC); # do accounting ...
                                setflag(FLT_ACCFAILED); # ... even if
the transaction fails
                        }
                        route(RELAY);
                } else {
                        if (is_method("SUBSCRIBE") && uri == myself) {
                                # in-dialog subscribe requests
                                route(PRESENCE);
                                exit;
                        }
                        if ( is_method("ACK") ) {
                                if ( t_check_trans() ) {
                                        # no loose-route, but stateful ACK;
                                        # must be an ACK after a 487
                                        # or e.g. 404 from upstream server
                                        t_relay();
                                        exit;
                                } else {
                                        # ACK without matching
transaction ... ignore and discard
                                        exit;
                                }
                        }
                        sl_send_reply("404","Not here");
                }
                exit;
        }
}

# Handle SIP registrations
route[REGISTRAR] {
        if (is_method("REGISTER"))
        {
                if(isflagset(FLT_NATS))
                {
                        setbflag(FLB_NATB);
                        # uncomment next line to do SIP NAT pinging
                        setbflag(FLB_NATSIPPING);
                }
                if (!save("location"))
                        sl_reply_error();

                exit;
        }
}

# USER location service
route[LOCATION] {

#!ifdef WITH_ALIASDB
        # search in DB-based aliases
        alias_db_lookup("dbaliases");
#!endif

        if (!lookup("location")) {
                switch ($rc) {
                        case -1:
                        case -3:
                                t_newtran();
                                t_reply("404", "Not Found");
                                exit;
                        case -2:
                                sl_send_reply("405", "Method Not Allowed");
                                exit;
                }
        }

        # when routing via usrloc, log the missed calls also
        if (is_method("INVITE"))
        {
                setflag(FLT_ACCMISSED);
        }
}

# Presence server route
route[PRESENCE] {
        if(!is_method("PUBLISH|SUBSCRIBE"))
                return;

#!ifdef WITH_PRESENCE
        if (!t_newtran())
        {
                sl_reply_error();
                exit;
        };

        if(is_method("PUBLISH"))
        {
                handle_publish();
                t_release();
        }
        else
        if( is_method("SUBSCRIBE"))
        {
                handle_subscribe();
                t_release();
        }
        exit;
#!endif

        # if presence enabled, this part will not be executed
        if (is_method("PUBLISH") || $rU==$null)
        {
                sl_send_reply("404", "Not here");
                exit;
        }
        return;
}

# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
        if (is_method("REGISTER"))
        {
                # authenticate the REGISTER requests (uncomment to enable auth)
                if (!www_authorize("$td", "subscriber"))
                {
                        www_challenge("$td", "0");
                        exit;
                }

                if ($au!=$tU)
                {
                        sl_send_reply("403","Forbidden auth ID");
                        exit;
                }
        } else {

#!ifdef WITH_IPAUTH
                if(allow_source_address())
                {
                        # source IP allowed
                        return;
                }
#!endif

                # authenticate if from local subscriber
                if (from_uri==myself)
                {
                        if (!proxy_authorize("$fd", "subscriber")) {
                                proxy_challenge("$fd", "0");
                                exit;
                        }
                        if (is_method("PUBLISH"))
                        {
                                if ($au!=$tU) {
                                        sl_send_reply("403","Forbidden
auth ID");
                                        exit;
                                }
                        } else {
                                if ($au!=$fU) {
                                        sl_send_reply("403","Forbidden
auth ID");
                                        exit;
                                }
                        }

                        consume_credentials();
                        # caller authenticated
                } else {
                        # caller is not local subscriber, then check if it calls
                        # a local destination, otherwise deny, not an
open relay here
                        if (!uri==myself)
                        {
                                sl_send_reply("403","Not relaying");
                                exit;
                        }
                }
        }
#!endif
        return;
}

# Caller NAT detection route
route[NAT] {
#!ifdef WITH_NAT
        force_rport();
        if (nat_uac_test("31")) {
                if (method=="REGISTER") {
                        fix_nated_register();
                } else {
                        fix_nated_contact();
                }
                setflag(FLT_NATS);
        }
#!endif
        return;
}

# RTPProxy control
route[RTPPROXY] {
#!ifdef WITH_NAT
        if (is_method("BYE")) {
                unforce_rtp_proxy();
        } else if (is_method("INVITE")){
                force_rtp_proxy("s");
        }
        if (!has_totag()) add_rr_param(";nat=yes");
#!endif
        return;
}

# Routing to foreign domains
route[SIPOUT] {
        if (!uri==myself)
        {
                append_hf("P-hint: outbound\r\n");
                route(RELAY);
        }
}

# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
        # check if PSTN GW IP is defined
        if (strempty($sel(cfg_get.pstn.gw_ip))) {
                xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not
defined\n");
                return;
        }

        # route to PSTN dialed numbers starting with '+' or '00'
        #     (international format)
        # - update the condition to match your dialing rules for PSTN routing
        if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
                return;

        # only local users allowed to call
        if(from_uri!=myself) {
                sl_send_reply("403", "Not Allowed");
                exit;
        }

        $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);

        route(RELAY);
        exit;
#!endif

        return;
}

# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC]
{
        # allow XMLRPC from localhost
        if ((method=="POST" || method=="GET")
                        && (src_ip==127.0.0.1)) {
                # close connection only for xmlrpclib user agents
(there is a bug in
                # xmlrpclib: it waits for EOF before interpreting the response).
                if ($hdr(User-Agent) =~ "xmlrpclib")
                        set_reply_close();
                set_reply_no_connect();
                dispatch_rpc();
                exit;
        }
        send_reply("403", "Forbidden");
        exit;
}
#!endif

# Sample branch router
branch_route[BRANCH_ONE] {
        xdbg("new branch at $ru\n");
}

# Sample onreply route
onreply_route[REPLY_ONE] {
        xdbg("incoming reply\n");
#!ifdef WITH_NAT
        if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))
                        && status=~"(183)|(2[0-9][0-9])") {
                force_rtp_proxy();
        }
        if (isbflagset("6")) {
                fix_nated_contact();
        }
#!endif
}

# Sample failure route
failure_route[FAIL_ONE] {
#!ifdef WITH_NAT
        if (is_method("INVITE")
                        && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) {
                unforce_rtp_proxy();
        }
#!endif

        if (t_is_canceled()) {
                exit;
        }

        # uncomment the following lines if you want to block client
        # redirect based on 3xx replies.
        ##if (t_check_status("3[0-9][0-9]")) {
        ##t_reply("404","Not found");
        ##      exit;
        ##}

        # uncomment the following lines if you want to redirect the failed
        # calls to a different new destination
        ##if (t_check_status("486|408")) {
        ##      sethostport("192.168.2.100:5060");
        ##      append_branch();
        ##      # do not set the missed call flag again
        ##      t_relay();
        ##}
}



On Mon, Jun 6, 2011 at 10:52 PM, Alex Balashov
<abalashov at evaristesys.com> wrote:
> What are the syntax issues?
>
> P.S. has_sdp() doesn't actually exist.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Jun 6, 2011, at 11:48 PM, Mokhtar Bengana <mokhtar.bengana at gmail.com> wrote:
>
>> Andrew,
>>
>> I am trying to use rtpproxy_offer/answer on the ONREPLY ROUTE but I am
>> having some syntax issues. Here is the example I am using. Any help
>> with the syntax appreciated.
>>
>>
>> route {
>> ...
>> ? ?if (is_method("INVITE")) {
>> ? ? ? ?if (has_sdp()) {
>> ? ? ? ? ? ?if (rtpproxy_offer())
>> ? ? ? ? ? ? ? ?t_on_reply("1");
>> ? ? ? ?} else {
>> ? ? ? ? ? ?t_on_reply("2");
>> ? ? ? ?}
>> ? ?}
>> ? ?if (is_method("ACK") && has_sdp())
>> ? ? ? ?rtpproxy_answer();
>> ...
>> }
>>
>> onreply_route[1]
>> {
>> ...
>> ? ?if (has_sdp())
>> ? ? ? ?rtpproxy_answer();
>> ...
>> }
>>
>> onreply_route[2]
>> {
>> ...
>> ? ?if (has_sdp())
>> ? ? ? ?rtpproxy_offer();
>> ...
>> }
>>
>> On Mon, Jun 6, 2011 at 5:37 AM, Andrew Pogrebennyk
>> <andrew.pogrebennyk at portaone.com> wrote:
>>> Mokhtar, could you please make sure that you are calling route(RTPPROXY) in
>>> reply route as well, as Alex suggested?
>>>
>>> On 03.06.2011 16:33, Mokhtar Bengana wrote:
>>>>
>>>> This is how I configured rtpproxy. Not sure why rtpproxy is not
>>>> engaged both ways. Thanks for your help.
>>>
>>> --
>>> Sincerely,
>>> Andrew Pogrebennyk
>>>
>>> _______________________________________________
>>> 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
>
> _______________________________________________
> 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 omar at 321communications.com  Wed Jun  8 13:10:32 2011
From: omar at 321communications.com (Omar)
Date: Wed, 8 Jun 2011 07:10:32 -0400
Subject: [SR-Users] 302 rewrite ruri based on contact
In-Reply-To: <BANLkTinUL-UopvEMBJROisVxybt2Hq7xPg@mail.gmail.com>
References: <AB1828DB-A9EB-4B10-9894-F87AFA3DBD6E@321communications.com>
	<BANLkTinUL-UopvEMBJROisVxybt2Hq7xPg@mail.gmail.com>
Message-ID: <1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>

Do i have to clear the variable in $var everytime a call ends a Bye or cancels, or the variable will clear it up automatically in Kamailio1.4.3? i think this acts as a vector, but i have some doubt regarding

$avp i think uses shared memory, then i have to take care to reserve enough memory for it in the shared memory right?


thanks

O


On May 24, 2011, at 6:58 PM, Ovidiu Sas wrote:

> Here's an example on how to perform forking on failure on 1.4:
> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4/examples/serial_183.cfg
> 
> 
> Regards,
> Ovidiu Sas
> 
> On Tue, May 24, 2011 at 4:30 PM, Omar <omar at 321communications.com> wrote:
>> 
>> I need to rewrite the destination after I receive 302
>> I have this logic
>> 
>> Onreply
>> Loads in avp the information needed
>> 
>> I try sending to failure route and to a route in both cases the $ru become <null>
>> 
>> In the route and failureroute
>> I did try rewritehost, rewriteuri, $ru= number+ host
>> And append the route
>> But $ru is always null.
>> Inside the route or failure route
>> 
>> If there are no 302 nor the onreply involved everything works fine
>> 
>> Any suggestion to rewrite the ruri after a 302 is welcome
>> 
>> I am still using Kamailo1.4.3
>> 
>> 
>> Omar
>> _______________________________________________
>> 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 osas at voipembedded.com  Wed Jun  8 15:15:11 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 8 Jun 2011 09:15:11 -0400
Subject: [SR-Users] 302 rewrite ruri based on contact
In-Reply-To: <1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>
References: <AB1828DB-A9EB-4B10-9894-F87AFA3DBD6E@321communications.com>
	<BANLkTinUL-UopvEMBJROisVxybt2Hq7xPg@mail.gmail.com>
	<1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>
Message-ID: <BANLkTikiuah0K-UUvbKHNw5fCaRS=aBkKg@mail.gmail.com>

$var are script variables (read more here):
http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#script_variables
Best thing to do is to initialize the variable every time before using it.
The variable is not transaction or call persistent (it's scope is in
context of the message being processed).

$avp may be transaction persistent (enable it in tm):
http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#avps
http://www.kamailio.org/docs/modules/1.4.x/tm.html#id2530425


Regards,
Ovidiu Sas


On Wed, Jun 8, 2011 at 7:10 AM, Omar <omar at 321communications.com> wrote:
> Do i have to clear the variable in $var everytime a call ends a Bye or cancels, or the variable will clear it up automatically in Kamailio1.4.3? i think this acts as a vector, but i have some doubt regarding
>
> $avp i think uses shared memory, then i have to take care to reserve enough memory for it in the shared memory right?
>
>
> thanks
>
> O
>
>
> On May 24, 2011, at 6:58 PM, Ovidiu Sas wrote:
>
>> Here's an example on how to perform forking on failure on 1.4:
>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4/examples/serial_183.cfg
>>
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Tue, May 24, 2011 at 4:30 PM, Omar <omar at 321communications.com> wrote:
>>>
>>> I need to rewrite the destination after I receive 302
>>> I have this logic
>>>
>>> Onreply
>>> Loads in avp the information needed
>>>
>>> I try sending to failure route and to a route in both cases the $ru become <null>
>>>
>>> In the route and failureroute
>>> I did try rewritehost, rewriteuri, $ru= number+ host
>>> And append the route
>>> But $ru is always null.
>>> Inside the route or failure route
>>>
>>> If there are no 302 nor the onreply involved everything works fine
>>>
>>> Any suggestion to rewrite the ruri after a 302 is welcome
>>>
>>> I am still using Kamailo1.4.3
>>>
>>>
>>> Omar
>>> _______________________________________________
>>> 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
>>
>
>
> _______________________________________________
> 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 henning.westerholt at 1und1.de  Wed Jun  8 16:00:15 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Wed, 8 Jun 2011 16:00:15 +0200
Subject: [SR-Users] 302 rewrite ruri based on contact
In-Reply-To: <1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>
References: <AB1828DB-A9EB-4B10-9894-F87AFA3DBD6E@321communications.com>
	<BANLkTinUL-UopvEMBJROisVxybt2Hq7xPg@mail.gmail.com>
	<1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>
Message-ID: <201106081600.15382.henning.westerholt@1und1.de>

On Wednesday 08 June 2011, Omar wrote:
> [..]
> $avp i think uses shared memory, then i have to take care to reserve enough
> memory for it in the shared memory right?

Hi Omar,

yes, AVPs live in shared memory as they are transactional. But you probably 
need to create a lot of variables until your shared memory pool will be empty. 
If you doubt that you've enough memory configured, just increase it - its only 
really used on the system when its actually filled with content.

Best regards,

Henning


From omar at 321communications.com  Wed Jun  8 16:01:27 2011
From: omar at 321communications.com (Omar)
Date: Wed, 8 Jun 2011 10:01:27 -0400
Subject: [SR-Users] 302 rewrite ruri based on contact
In-Reply-To: <BANLkTikiuah0K-UUvbKHNw5fCaRS=aBkKg@mail.gmail.com>
References: <AB1828DB-A9EB-4B10-9894-F87AFA3DBD6E@321communications.com>
	<BANLkTinUL-UopvEMBJROisVxybt2Hq7xPg@mail.gmail.com>
	<1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>
	<BANLkTikiuah0K-UUvbKHNw5fCaRS=aBkKg@mail.gmail.com>
Message-ID: <10EE12D1-5274-4FED-A391-AE8E49A21120@321communications.com>

Thanks ovidiu

O

On Jun 8, 2011, at 9:15 AM, Ovidiu Sas wrote:

> $var are script variables (read more here):
> http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#script_variables
> Best thing to do is to initialize the variable every time before using it.
> The variable is not transaction or call persistent (it's scope is in
> context of the message being processed).
> 
> $avp may be transaction persistent (enable it in tm):
> http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#avps
> http://www.kamailio.org/docs/modules/1.4.x/tm.html#id2530425
> 
> 
> Regards,
> Ovidiu Sas
> 
> 
> On Wed, Jun 8, 2011 at 7:10 AM, Omar <omar at 321communications.com> wrote:
>> Do i have to clear the variable in $var everytime a call ends a Bye or cancels, or the variable will clear it up automatically in Kamailio1.4.3? i think this acts as a vector, but i have some doubt regarding
>> 
>> $avp i think uses shared memory, then i have to take care to reserve enough memory for it in the shared memory right?
>> 
>> 
>> thanks
>> 
>> O
>> 
>> 
>> On May 24, 2011, at 6:58 PM, Ovidiu Sas wrote:
>> 
>>> Here's an example on how to perform forking on failure on 1.4:
>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4/examples/serial_183.cfg
>>> 
>>> 
>>> Regards,
>>> Ovidiu Sas
>>> 
>>> On Tue, May 24, 2011 at 4:30 PM, Omar <omar at 321communications.com> wrote:
>>>> 
>>>> I need to rewrite the destination after I receive 302
>>>> I have this logic
>>>> 
>>>> Onreply
>>>> Loads in avp the information needed
>>>> 
>>>> I try sending to failure route and to a route in both cases the $ru become <null>
>>>> 
>>>> In the route and failureroute
>>>> I did try rewritehost, rewriteuri, $ru= number+ host
>>>> And append the route
>>>> But $ru is always null.
>>>> Inside the route or failure route
>>>> 
>>>> If there are no 302 nor the onreply involved everything works fine
>>>> 
>>>> Any suggestion to rewrite the ruri after a 302 is welcome
>>>> 
>>>> I am still using Kamailo1.4.3
>>>> 
>>>> 
>>>> Omar
>>>> _______________________________________________
>>>> 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
>>> 
>> 
>> 
>> _______________________________________________
>> 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 skodonnell at ureach.com  Wed Jun  8 18:28:03 2011
From: skodonnell at ureach.com (Sean O'Donnell)
Date: Wed, 8 Jun 2011 12:28:03 -0400
Subject: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
Message-ID: <201106081628.p58GS3hn029311@ureach.com>

Hi:

We're running kamailio 1.5.5 as part of our VoIP infrastructure.  One of our
service providers
has an SBC that sends an OPTIONS messages with a Max-Forwards of 0.

Looking at RFC 3261 section 11, this seems to be a valid method of forcing the
proxy to respond to the OPTIONS.
However, the standard initial sanity checks in most proxy configs will reject
this message.  I fixed our script
by changing to this:
  
        # initial sanity checks -- messages with
        # max_forwards==0, or excessively long requests
        if (!mf_process_maxfwd_header("10")) {
                if( method != "OPTIONS" ) {
                        sl_send_reply("483","Too Many Hops");
                        return;
                };
        };
        if ( msg:len > max_len ) {
                sl_send_reply("513", "Message too big");
                return;
        };

but does anyone have a better solution?

Thanks,

Sean O'Donnell



From agranig at sipwise.com  Wed Jun  8 18:49:09 2011
From: agranig at sipwise.com (Andreas Granig)
Date: Wed, 08 Jun 2011 18:49:09 +0200
Subject: [SR-Users] Incorrect registrar Expires handling
In-Reply-To: <4DE80BC1.3090903@sipwise.com>
References: <4DE7E343.1090509@sipwise.com>
	<4DE7F17C.7070709@gmail.com>	<4DE7F743.4090501@sipwise.com>
	<4DE80A43.5000706@gmail.com> <4DE80BC1.3090903@sipwise.com>
Message-ID: <4DEFA805.20105@sipwise.com>

Hi all,

Just wanted to confirm that commit
16183d4553aca270da1284255f3f6862132a3aeb in branch 3.1 fixes the problem
described below.

Thanks Daniel for you prompt help!

Andreas

On 06/03/2011 12:16 AM, Andreas Granig wrote:
> Hi,
> 
> On 06/03/2011 12:10 AM, Daniel-Constantin Mierla wrote:
>> one more thing, what is the output of kamctl ul show? I want to see if
>> it is a matter of building the contact header for reply or is in
>> internal list of location records.
> 
> Domain:: location table=512 records=1 max_slot=1
> 	AOR:: me at somedomain.com
> 		Contact:: sip:me at 10.0.0.7:52030;ob Q=
> 			Expires:: 875
> 			Callid:: DZZHTDttFd5WivydjNnWfpRLsXp8RIYh
> 			Cseq:: 4599
> 			User-agent:: myclient
> 			Received:: sip:1.2.3.4:55642
> 			Path:: <sip:lb at 127.0.0.1:5060;lr;received=sip:1.2.3.4:55642>
> 			State:: CS_SYNC
> 			Flags:: 0
> 			Cflag:: 192
> 			Socket:: udp:127.0.0.1:5062
> 			Methods:: 8143
> 		Contact:: sip:me at 1.2.3.4:55642;transport=UDP;ob Q=
> 			Expires:: 876
> 			Callid:: DZZHTDttFd5WivydjNnWfpRLsXp8RIYh
> 			Cseq:: 4601
> 			User-agent:: myclient
> 			Path:: <sip:lb at 127.0.0.1:5060;lr>
> 			State:: CS_SYNC
> 			Flags:: 0
> 			Cflag:: 0
> 			Socket:: udp:127.0.0.1:5062
> 			Methods:: 8143
> 
> Hope this helps,
> 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 klaus.mailinglists at pernau.at  Wed Jun  8 20:08:48 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 08 Jun 2011 20:08:48 +0200
Subject: [SR-Users] Best way to handle OPTIONS message with Max-Forwards
 of 0
In-Reply-To: <201106081628.p58GS3hn029311@ureach.com>
References: <201106081628.p58GS3hn029311@ureach.com>
Message-ID: <4DEFBAB0.7080009@pernau.at>

What is the problem with the default config?

The proxy will respond with 483 as the standard defines. I guess the SBC
will doe this to check if the proxy is alive and will accept any
response as "the proxy is alive".

regards
Klaus


Am 08.06.2011 18:28, schrieb Sean O'Donnell:
> Hi:
> 
> We're running kamailio 1.5.5 as part of our VoIP infrastructure.  One of our
> service providers
> has an SBC that sends an OPTIONS messages with a Max-Forwards of 0.
> 
> Looking at RFC 3261 section 11, this seems to be a valid method of forcing the
> proxy to respond to the OPTIONS.
> However, the standard initial sanity checks in most proxy configs will reject
> this message.  I fixed our script
> by changing to this:
>   
>         # initial sanity checks -- messages with
>         # max_forwards==0, or excessively long requests
>         if (!mf_process_maxfwd_header("10")) {
>                 if( method != "OPTIONS" ) {
>                         sl_send_reply("483","Too Many Hops");
>                         return;
>                 };
>         };
>         if ( msg:len > max_len ) {
>                 sl_send_reply("513", "Message too big");
>                 return;
>         };
> 
> but does anyone have a better solution?
> 
> Thanks,
> 
> Sean O'Donnell
> 
> 
> _______________________________________________
> 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  Wed Jun  8 20:22:37 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 08 Jun 2011 20:22:37 +0200
Subject: [SR-Users] Best way to handle OPTIONS message with Max-Forwards
 of 0
In-Reply-To: <201106081628.p58GS3hn029311@ureach.com>
References: <201106081628.p58GS3hn029311@ureach.com>
Message-ID: <4DEFBDED.20007@pernau.at>

Anyway - not standard conform but efficient (put somewhere in the beginning)

force_rport();
if (is_method("OPTIONS")) {
    sl_send_reply("200","OK");
    exit;
}
# sipura/linksys keepalive
if (is_method("NOTIFY")) {
    if ( search("^Event: keep-alive") ) {
            sl_send_reply("200","Keep Alive");
            exit;
    }
}
# ignore requests generated by sipvicious
# User-Agent: friendly-scanner
if ($ua == "friendly-scanner") {
        exit;
}


regards
Klaus

Am 08.06.2011 18:28, schrieb Sean O'Donnell:
> Hi:
> 
> We're running kamailio 1.5.5 as part of our VoIP infrastructure.  One of our
> service providers
> has an SBC that sends an OPTIONS messages with a Max-Forwards of 0.
> 
> Looking at RFC 3261 section 11, this seems to be a valid method of forcing the
> proxy to respond to the OPTIONS.
> However, the standard initial sanity checks in most proxy configs will reject
> this message.  I fixed our script
> by changing to this:
>   
>         # initial sanity checks -- messages with
>         # max_forwards==0, or excessively long requests
>         if (!mf_process_maxfwd_header("10")) {
>                 if( method != "OPTIONS" ) {
>                         sl_send_reply("483","Too Many Hops");
>                         return;
>                 };
>         };
>         if ( msg:len > max_len ) {
>                 sl_send_reply("513", "Message too big");
>                 return;
>         };
> 
> but does anyone have a better solution?
> 
> Thanks,
> 
> Sean O'Donnell
> 
> 
> _______________________________________________
> 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 skodonnell at ureach.com  Wed Jun  8 21:49:06 2011
From: skodonnell at ureach.com (Sean O'Donnell)
Date: Wed, 8 Jun 2011 15:49:06 -0400
Subject: [SR-Users] Best way to handle OPTIONS message with Max-Forwards
	of 0
Message-ID: <201106081949.p58Jn6N1005059@ureach.com>

Thanks for the reply.

In our case, the SBC treated the 483 as an error of some sort and wouldn't send
traffic.  It wanted a 200/OK.

Anyway, I thought that when a proxy responds to an OPTIONS, its supposed to
answer with a 200/OK (3261 11.2).  Is that correct?

Sean O'Donnell




---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists at pernau.at) wrote:

What is the problem with the default config?

The proxy will respond with 483 as the standard defines. I guess the SBC
will doe this to check if the proxy is alive and will accept any
response as "the proxy is alive".

regards
Klaus


Am 08.06.2011 18:28, schrieb Sean O'Donnell:
> Hi:
> 
> We're running kamailio 1.5.5 as part of our VoIP infrastructure.  One of our
> service providers
> has an SBC that sends an OPTIONS messages with a Max-Forwards of 0.
> 
> Looking at RFC 3261 section 11, this seems to be a valid method of forcing the
> proxy to respond to the OPTIONS.
> However, the standard initial sanity checks in most proxy configs will reject
> this message.  I fixed our script
> by changing to this:
>   
>         # initial sanity checks -- messages with
>         # max_forwards==0, or excessively long requests
>         if (!mf_process_maxfwd_header("10")) {
>                 if( method != "OPTIONS" ) {
>                         sl_send_reply("483","Too Many Hops");
>                         return;
>                 };
>         };
>         if ( msg:len > max_len ) {
>                 sl_send_reply("513", "Message too big");
>                 return;
>         };
> 
> but does anyone have a better solution?
> 
> Thanks,
> 
> Sean O'Donnell
> 
> 
> _______________________________________________
> 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 klaus.mailinglists at pernau.at  Wed Jun  8 22:03:38 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 08 Jun 2011 22:03:38 +0200
Subject: [SR-Users] Best way to handle OPTIONS message with Max-Forwards
 of 0
In-Reply-To: <201106081949.p58Jn6N1005059@ureach.com>
References: <201106081949.p58Jn6N1005059@ureach.com>
Message-ID: <4DEFD59A.7070809@pernau.at>

IMO RFC 3261 does not cleary state that this OPTIONS request should be
answered with 200 OK.

Anyway, if the SBC wants 200 OK, then send 200 OK :-)

klaus

Am 08.06.2011 21:49, schrieb Sean O'Donnell:
> Thanks for the reply.
> 
> In our case, the SBC treated the 483 as an error of some sort and wouldn't send
> traffic.  It wanted a 200/OK.
> 
> Anyway, I thought that when a proxy responds to an OPTIONS, its supposed to
> answer with a 200/OK (3261 11.2).  Is that correct?
> 
> Sean O'Donnell
> 
> 
> 
> 
> ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists at pernau.at) wrote:
> 
> What is the problem with the default config?
> 
> The proxy will respond with 483 as the standard defines. I guess the SBC
> will doe this to check if the proxy is alive and will accept any
> response as "the proxy is alive".
> 
> regards
> Klaus
> 
> 
> Am 08.06.2011 18:28, schrieb Sean O'Donnell:
>> Hi:
>>
>> We're running kamailio 1.5.5 as part of our VoIP infrastructure.  One of our
>> service providers
>> has an SBC that sends an OPTIONS messages with a Max-Forwards of 0.
>>
>> Looking at RFC 3261 section 11, this seems to be a valid method of forcing the
>> proxy to respond to the OPTIONS.
>> However, the standard initial sanity checks in most proxy configs will reject
>> this message.  I fixed our script
>> by changing to this:
>>   
>>         # initial sanity checks -- messages with
>>         # max_forwards==0, or excessively long requests
>>         if (!mf_process_maxfwd_header("10")) {
>>                 if( method != "OPTIONS" ) {
>>                         sl_send_reply("483","Too Many Hops");
>>                         return;
>>                 };
>>         };
>>         if ( msg:len > max_len ) {
>>                 sl_send_reply("513", "Message too big");
>>                 return;
>>         };
>>
>> but does anyone have a better solution?
>>
>> Thanks,
>>
>> Sean O'Donnell
>>
>>
>> _______________________________________________
>> 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
> 
> 
> 
> 
> _______________________________________________
> 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 Eugen.Dedu at pu-pm.univ-fcomte.fr  Wed Jun  8 22:06:23 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Wed, 08 Jun 2011 22:06:23 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
Message-ID: <4DEFD63F.5080109@pu-pm.univ-fcomte.fr>

No idea?

On 05/06/11 22:31, Eugen Dedu wrote:
> Hi,
>
> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
> users who are not registered an empty NOTIFY body is returned when asked
> by a SUBSCRIBE. What does this mean from SIP standard point of view, and
> from kamailio point of view (are they identical?) I see in RFC3265/3.1.6.2:
> .... If the resource
> has no meaningful state at the time that the SUBSCRIBE message is
> processed, this NOTIFY message MAY contain an empty or neutral body
> but is difficult for me to interpret what it means.
>
> Example: I ask the presence for a user xyz who registered and quit
> application long time ago:
>
> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
> CSeq: 1 SUBSCRIBE
> Via: SIP/2.0/UDP
> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport
>
> User-Agent: Ekiga/3.3.1
> From: <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
> Supported: eventlist
> To: <sip:xyz at ekiga.net>
> Accept: application/pidf+xml
> Accept: multipart/related
> Accept: application/rlmi+xml
> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
>
> Expires: 300
> Event: presence
> Content-Length: 0
> Max-Forwards: 70
>
> I receive the following answer:
>
> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
> CSeq: 1 NOTIFY
> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
> User-Agent: Kamailio (1.5.3-notls (i386/linux))
> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
> Contact: <sip:86.64.162.35:5060>
> Subscription-State: active;expires=370
> Event: presence
> Content-Length: 0
> Max-Forwards: 70
>
> To resume: What does SIP standard say about this NOTIFY with empty body?
> Does this mean that the user xyz is offline?
>
> Or does this mean that user's status has not changed? In fact, the
> NOTIFY with empty body (as shown above) is the first one sent by
> kamailio, so there is no "previous" state of that user, hence
> "unchanged" status has no meaning.
>
> Thank you,


From skodonnell at ureach.com  Thu Jun  9 04:59:46 2011
From: skodonnell at ureach.com (Sean O'Donnell)
Date: Wed, 8 Jun 2011 22:59:46 -0400
Subject: [SR-Users] Best way to handle OPTIONS message with Max-Forwards
	of 0
Message-ID: <201106090259.p592xkMC017040@ureach.com>

Looks more comprehensive than mine.  Thanks!

Sean O'Donnell




---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists at pernau.at) wrote:

Anyway - not standard conform but efficient (put somewhere in the beginning)

force_rport();
if (is_method("OPTIONS")) {
    sl_send_reply("200","OK");
    exit;
}
# sipura/linksys keepalive
if (is_method("NOTIFY")) {
    if ( search("^Event: keep-alive") ) {
            sl_send_reply("200","Keep Alive");
            exit;
    }
}
# ignore requests generated by sipvicious
# User-Agent: friendly-scanner
if ($ua == "friendly-scanner") {
        exit;
}


regards
Klaus

Am 08.06.2011 18:28, schrieb Sean O'Donnell:
> Hi:
> 
> We're running kamailio 1.5.5 as part of our VoIP infrastructure.  One of our
> service providers
> has an SBC that sends an OPTIONS messages with a Max-Forwards of 0.
> 
> Looking at RFC 3261 section 11, this seems to be a valid method of forcing the
> proxy to respond to the OPTIONS.
> However, the standard initial sanity checks in most proxy configs will reject
> this message.  I fixed our script
> by changing to this:
>   
>         # initial sanity checks -- messages with
>         # max_forwards==0, or excessively long requests
>         if (!mf_process_maxfwd_header("10")) {
>                 if( method != "OPTIONS" ) {
>                         sl_send_reply("483","Too Many Hops");
>                         return;
>                 };
>         };
>         if ( msg:len > max_len ) {
>                 sl_send_reply("513", "Message too big");
>                 return;
>         };
> 
> but does anyone have a better solution?
> 
> Thanks,
> 
> Sean O'Donnell
> 
> 
> _______________________________________________
> 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 ibc at aliax.net  Thu Jun  9 12:44:11 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 12:44:11 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
Message-ID: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>

Hi, I'm testing Kamailio's NAPTR resolution:

version: kamailio 3.2.0-dev4 (x86_64/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, DBG_QM_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


DNS related configuration:

dns_servers_no=2
dns_srv_lb=yes
dns_try_naptr=yes
dns_use_search_list=no
use_dns_cache=yes
use_dns_failover=yes
dns_tls_preference=1
dns_tcp_preference=1
dns_udp_preference=1


As the doc says it should respect the priorities of the NAPTR records,
see http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#dns_sctp_pref_dns_tcp_pref_dns_tls_pref_dns_udp_pref:

  "To use the remote site preferences set all dns_*_pref to the same
positive value
    (e.g. dns_udp_pref=1, dns_tcp_pref=1, dns_tls_pref=1, dns_sctp_pref=1)"


So my kamailio receives an INVITE for an external domain oversip.net
(no transport param neither port in the RURI) and does a t_relay.
Request RURI is sip:qwe at oversip.net.

According to NAPTR:

  ~$ host -t naptr oversip.net
  oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net.
  oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net.
  oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net.
  oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net.
  oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net.

So it should try TLS over TCP first, if it fails try TCP and if it
fails try UDP.


However it just uses UDP, why??
Even if I set a minor value to dns_tls_preference (so higher priority
I expect) it still uses UDP.

Am I doing something wrong? Thanks a lot.


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


From ibc at aliax.net  Thu Jun  9 12:51:19 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 12:51:19 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
Message-ID: <BANLkTikJioTa2_b7A3uCE2U2cEHZbibmUQ@mail.gmail.com>

2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> According to NAPTR:
>
> ?~$ host -t naptr oversip.net
> ?oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net.
> ?oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net.
> ?oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net.
> ?oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net.
> ?oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net.
>
> So it should try TLS over TCP first, if it fails try TCP and if it
> fails try UDP.

Just to confirm, the above NAPTR record for "SIPS+D2T" has order 5 and
preference 50. According to RFC 2915:

   Order
      A 16-bit unsigned integer specifying the order in which the NAPTR
      records MUST be processed to ensure the correct ordering of
      rules.  Low numbers are processed before high numbers

   Preference
      A 16-bit unsigned integer that specifies the order in which NAPTR
      records with equal "order" values SHOULD be processed, low
      numbers being processed before high numbers.


So as my domain oversip.net has no entries with same order, preference
value doesn't matter. And of course, SIP over TLS should take
preference.

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


From ibc at aliax.net  Thu Jun  9 12:54:01 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 12:54:01 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTikJioTa2_b7A3uCE2U2cEHZbibmUQ@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTikJioTa2_b7A3uCE2U2cEHZbibmUQ@mail.gmail.com>
Message-ID: <BANLkTikWeseZUsyWtt_Wn5+2po3UejkEJA@mail.gmail.com>

2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> So as my domain oversip.net has no entries with same order, preference
> value doesn't matter. And of course, SIP over TLS should take
> preference.

Also my SRV records are ok (using other SIP clients they choose TLS first):


$ host -t srv _sips._tcp.oversip.net.
_sips._tcp.oversip.net has SRV record 1 80 9091 sip1.oversip.net.

$ host -t a sip1.oversip.net.
sip1.oversip.net has address 91.121.79.216


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


From ibc at aliax.net  Thu Jun  9 13:08:35 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 13:08:35 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
Message-ID: <BANLkTikud9FccV+k_j6MAO4nAE0QULykdw@mail.gmail.com>

2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> According to NAPTR:
>
> ?~$ host -t naptr oversip.net
> ?oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net.
> ?oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net.
> ?oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net.
> ?oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net.
> ?oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net.
>
> So it should try TLS over TCP first, if it fails try TCP and if it
> fails try UDP.
>
>
> However it just uses UDP, why??


This is the Kamailio's log about DNS resolution for the above request
(note that I've restarted kamailio before the request in order to
clean any possible DNS cache):


DEBUG: <core> [dns_cache.c:567]: dns_hash_find(oversip.net(11), 35), h=956
DEBUG: <core> [resolve.c:924]: get_record: skipping 0 NS (p=0x826fdd,
end=0x826fdd)
DEBUG: <core> [resolve.c:940]: get_record: parsing 0 ARs (p=0x826fdd,
end=0x826fdd)
DEBUG: <core> [dns_cache.c:1777]: dns_get_related(0x7f359f68db50
(oversip.net, 35), 35, *(nil)) (0)
DEBUG: <core> [dns_cache.c:870]: dns_cache_add: adding oversip.net(11)
35 (flags=0) at 956
DEBUG: <core> [dns_cache.c:567]:
dns_hash_find(_sip._udp.oversip.net(21), 33), h=23
DEBUG: <core> [resolve.c:924]: get_record: skipping 0 NS (p=0x826f0a,
end=0x826f0a)
DEBUG: <core> [resolve.c:940]: get_record: parsing 0 ARs (p=0x826f0a,
end=0x826f0a)
DEBUG: <core> [dns_cache.c:1777]: dns_get_related(0x7f359f68de08
(_sip._udp.oversip.net, 33), 33, *(nil)) (0)
DEBUG: <core> [dns_cache.c:870]: dns_cache_add: adding
_sip._udp.oversip.net(21) 33 (flags=0) at 23
DEBUG: <core> [dns_cache.c:2362]: dns_srv_get_nxt_rr(0x7f359f68de08,
0, 0, 1367445037): selected 0/1 in grp. 0 (rand_w=0, rr=0x7f359f68de60
p=0 w=0 rsum=0)
DEBUG: <core> [dns_cache.c:567]: dns_hash_find(sip.oversip.net(15), 1), h=400
DEBUG: <core> [resolve.c:924]: get_record: skipping 0 NS (p=0x826ef1,
end=0x826ef1)
DEBUG: <core> [resolve.c:940]: get_record: parsing 0 ARs (p=0x826ef1,
end=0x826ef1)
DEBUG: <core> [dns_cache.c:1777]: dns_get_related(0x7f359f68def0
(sip.oversip.net, 1), 1, *(nil)) (0)
DEBUG: <core> [dns_cache.c:870]: dns_cache_add: adding
sip.oversip.net(15) 1 (flags=0) at 400
DEBUG: <core> [dns_cache.c:2982]: dns_a_resovle(sip.oversip.net, 0) returning 0
DEBUG: <core> [dns_cache.c:3236]:
dns_srv_resolve_ip("_sip._udp.oversip.net", 0, 0), ret=0,
ip=91.121.79.216
DEBUG: <core> [dns_cache.c:3358]: dns_sip_resolve(oversip.net, 0, 0),
srv0, ret=0


As per these logs I understand that no NAPTR record is taking place,
but just SRV for _sip._udp.oversip.net.



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


From miconda at gmail.com  Thu Jun  9 13:10:19 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 09 Jun 2011 13:10:19 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
Message-ID: <4DF0AA1B.9060408@gmail.com>

Hello,

the notify is sent to inform about the state of the subscription, which 
is active in this case. If it would the first subscription to that user 
and force_active will not be set, then should be subscription state 
pending, iirc.

The empty body does not change anything to the phone information about 
the watched presentity, which was known to be offline.

Cheers,
Daniel

On 6/8/11 10:06 PM, Eugen Dedu wrote:
> No idea?
>
> On 05/06/11 22:31, Eugen Dedu wrote:
>> Hi,
>>
>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>> users who are not registered an empty NOTIFY body is returned when asked
>> by a SUBSCRIBE. What does this mean from SIP standard point of view, and
>> from kamailio point of view (are they identical?) I see in 
>> RFC3265/3.1.6.2:
>> .... If the resource
>> has no meaningful state at the time that the SUBSCRIBE message is
>> processed, this NOTIFY message MAY contain an empty or neutral body
>> but is difficult for me to interpret what it means.
>>
>> Example: I ask the presence for a user xyz who registered and quit
>> application long time ago:
>>
>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>> CSeq: 1 SUBSCRIBE
>> Via: SIP/2.0/UDP
>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport 
>>
>>
>> User-Agent: Ekiga/3.3.1
>> From: <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>> Supported: eventlist
>> To: <sip:xyz at ekiga.net>
>> Accept: application/pidf+xml
>> Accept: multipart/related
>> Accept: application/rlmi+xml
>> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
>> Allow:
>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK 
>>
>>
>> Expires: 300
>> Event: presence
>> Content-Length: 0
>> Max-Forwards: 70
>>
>> I receive the following answer:
>>
>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>> CSeq: 1 NOTIFY
>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>> Contact: <sip:86.64.162.35:5060>
>> Subscription-State: active;expires=370
>> Event: presence
>> Content-Length: 0
>> Max-Forwards: 70
>>
>> To resume: What does SIP standard say about this NOTIFY with empty body?
>> Does this mean that the user xyz is offline?
>>
>> Or does this mean that user's status has not changed? In fact, the
>> NOTIFY with empty body (as shown above) is the first one sent by
>> kamailio, so there is no "previous" state of that user, hence
>> "unchanged" status has no meaning.
>>
>> 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



From ibc at aliax.net  Thu Jun  9 14:13:09 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 14:13:09 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
Message-ID: <BANLkTinvGjxc=PXHRAMTb7ZowmZQubR_Fg@mail.gmail.com>

2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> Even if I set a minor value to dns_tls_preference (so higher priority
> I expect) it still uses UDP.

By reading the doc it seems that higher values of dns_xxx_preference
take preference (it's the opposite to NAPTR records "order"). However
as I said I'm trying setting the same value for all the transports.

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


From miconda at gmail.com  Thu Jun  9 14:44:14 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 09 Jun 2011 14:44:14 +0200
Subject: [SR-Users] New developer: Peter Dunkley
Message-ID: <4DF0C01E.4060809@gmail.com>

Hello,

just to let everyone to know that starting with today Peter got 
developer access to GIT repository. He sent many patches in the past 
months to fix issues or add new features to presence modules (e.g., OMA 
extensions to embedded XCAP server) -- some are pending to be committed 
and expect more to come, as he is continuing to work with this part of 
SIP server.

The user id is 'pd'.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Thu Jun  9 15:26:36 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 09 Jun 2011 15:26:36 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTinvGjxc=PXHRAMTb7ZowmZQubR_Fg@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTinvGjxc=PXHRAMTb7ZowmZQubR_Fg@mail.gmail.com>
Message-ID: <4DF0CA0C.5000806@gmail.com>

Hi Inaki,

I kind of lost the conclusion on your replies, is it not working as 
documented/expected, or still not?

Cheers,
Daniel

On 6/9/11 2:13 PM, I?aki Baz Castillo wrote:
> 2011/6/9 I?aki Baz Castillo<ibc at aliax.net>:
>> Even if I set a minor value to dns_tls_preference (so higher priority
>> I expect) it still uses UDP.
> By reading the doc it seems that higher values of dns_xxx_preference
> take preference (it's the opposite to NAPTR records "order"). However
> as I said I'm trying setting the same value for all the transports.

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Thu Jun  9 16:11:15 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 16:11:15 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF0CA0C.5000806@gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTinvGjxc=PXHRAMTb7ZowmZQubR_Fg@mail.gmail.com>
	<4DF0CA0C.5000806@gmail.com>
Message-ID: <BANLkTi=5n6c_8rKuC2yFk_N_ufoPd3WGow@mail.gmail.com>

2011/6/9 Daniel-Constantin Mierla <miconda at gmail.com>:
> I kind of lost the conclusion on your replies, is it not working as
> documented/expected, or still not?

Sorry for so many mails. No, it doesn't work as documented or
expected. Summarizing:

My Kamailio is not performing NAPTR query for a Request URI
"sip:xxxx at oversip.net" (no transport, no port). It just does SRV query
for SIP over UDP.


Kamailio info:

--------------------
version: kamailio 3.2.0-dev5 (x86_64/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, DBG_QM_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 04:43:00 Jun  9 2011 with gcc 4.4.3
-------------------


A MESSAGE with RURI "sip:qwe at oversip.net" arrives (no transport param,
no port), t_relay() is invoked, and kamailio just performs a SRV query
for SIP over UDP (as logs show):

-----------------------
DEBUG: <core> [dns_cache.c:567]: dns_hash_find(oversip.net(11), 35), h=956
DEBUG: <core> [dns_cache.c:1777]: dns_get_related(0x7f98eebec390
(oversip.net, 35), 35, *(nil)) (0)
DEBUG: <core> [dns_cache.c:870]: dns_cache_add: adding oversip.net(11)
35 (flags=0) at 956
DEBUG: <core> [dns_cache.c:567]:
dns_hash_find(_sip._udp.oversip.net(21), 33), h=23
DEBUG: <core> [dns_cache.c:1777]: dns_get_related(0x7f98eebec648
(_sip._udp.oversip.net, 33), 33, *(nil)) (0)
DEBUG: <core> [dns_cache.c:870]: dns_cache_add: adding
_sip._udp.oversip.net(21) 33 (flags=0) at 23
DEBUG: <core> [dns_cache.c:2362]: dns_srv_get_nxt_rr(0x7f98eebec648,
0, 0, 1616155624): selected 0/1 in grp. 0 (rand_w=0, rr=0x7f98eebec6a0
p=0 w=0 rsum=0)
DEBUG: <core> [dns_cache.c:567]: dns_hash_find(sip.oversip.net(15), 1), h=400
DEBUG: <core> [dns_cache.c:1777]: dns_get_related(0x7f98eebec730
(sip.oversip.net, 1), 1, *(nil)) (0)
DEBUG: <core> [dns_cache.c:870]: dns_cache_add: adding
sip.oversip.net(15) 1 (flags=0) at 400
DEBUG: <core> [dns_cache.c:2982]: dns_a_resovle(sip.oversip.net, 0) returning 0
DEBUG: <core> [dns_cache.c:3236]:
dns_srv_resolve_ip("_sip._udp.oversip.net", 0, 0), ret=0,
ip=91.121.79.216
DEBUG: <core> [dns_cache.c:3358]: dns_sip_resolve(oversip.net, 0, 0),
srv0, ret=0
------------------------



My DNS related configuration in Kamailio:

-----------------------
dns_try_ipv6=no
dns_retr_time=1
dns_retr_no=2
dns_servers_no=2
dns_srv_lb=yes
dns_try_naptr=yes
dns_use_search_list=no
dns_search_full_match=no
use_dns_cache=yes
dns_cache_init=yes
use_dns_failover=yes
dns_tls_preference=1
dns_tcp_preference=1
dns_udp_preference=1
-----------------------


According to the doc it should perform NAPTR query (it's compiled with
USE_NAPTR as "kamailio -V" says, and parameter dns_try_naptr has value
"yes").


Could somebody try to send a MESSAGE or INVITE through Kamailio 3.X
with RURI sip:whatever at oversip.net? According to NAPTR/SRV records,
destinations should be (in order of preference):

1) TLS  91.121.79.216:9091
2) TCP  91.121.79.216:9090
3) UDP  91.121.79.216:9090



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


From alex at speakup.nl  Thu Jun  9 20:26:02 2011
From: alex at speakup.nl (Alex Hermann)
Date: Thu, 9 Jun 2011 20:26:02 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
Message-ID: <201106092026.02795.alex@speakup.nl>

On Thursday 09 June 2011 12:44:11 I?aki Baz Castillo wrote:
> 
> According to NAPTR:
> 
>   ~$ host -t naptr oversip.net
>   oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" ""
> _sips._tcp.oversip.net. oversip.net has NAPTR record 10 50 "S" "SIP+D2T"
> "" _sip._tcp.oversip.net. oversip.net has NAPTR record 20 50 "S" "SIP+D2U"
> "" _sip._udp.oversip.net. oversip.net has NAPTR record 40 50 "S" "SIP+D2S"
> "" _sip._sctp.oversip.net. oversip.net has NAPTR record 50 50 "S"
> "SIPS+D2S" "" _sips._sctp.oversip.net.
> 
> So it should try TLS over TCP first, if it fails try TCP and if it
> fails try UDP.
> 
> 
> However it just uses UDP, why??
> Even if I set a minor value to dns_tls_preference (so higher priority
> I expect) it still uses UDP.


The way I read rfc2915, there is no failover mechanism. The application pick 
the first target that it supports and uses that. There is no mention of trying 
other records afterwards. Matching/finding NAPTR records stops once the first 
match is completed. All other records are discarded. From  section 2:

   Order
      A 16-bit unsigned integer specifying the order in which the NAPTR
      records MUST be processed to ensure the correct ordering of
      rules.  Low numbers are processed before high numbers, and once a
      NAPTR is found whose rule "matches" the target, the client MUST
      NOT consider any NAPTRs with a higher value for order (except as
      noted below for the Flags field).

Note the last sentence.

-- 
Alex Hermann


From nangel at nothome.org  Thu Jun  9 19:00:47 2011
From: nangel at nothome.org (Nathan Angelacos)
Date: Thu, 09 Jun 2011 13:00:47 -0400
Subject: [SR-Users] Functions for DNS
Message-ID: <4DF0FC3F.7010300@nothome.org>

I'm looking to build a set of contacts with q values based on SRV 
records for serial/parallel forking.

I want "enum_query" (loading the contact set with a q value based on the 
order,preference of the NAPTR record)  but the ruri is not an e.164 
number, and SRV records are used instead.   Think in terms of "find all 
peer proxies based on SRV lookup, and sort them for use with 
t_load_contacts()/t_next_contacts()."

So far I haven't found any api that exposes the DNS lookups in sip-router.

Before rolling something custom with Lua, am I missing something that's 
already documented?   Or are there plans to make such a module?  Sure 
don't want to re-invent the wheel.

tia



From ibc at aliax.net  Thu Jun  9 19:03:33 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 19:03:33 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <201106092026.02795.alex@speakup.nl>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<201106092026.02795.alex@speakup.nl>
Message-ID: <BANLkTi=ijY08owMepH4K2yDrKQrNfQZzeQ@mail.gmail.com>

2011/6/9 Alex Hermann <alex at speakup.nl>:
> The way I read rfc2915, there is no failover mechanism. The application pick
> the first target that it supports and uses that. There is no mention of trying
> other records afterwards. Matching/finding NAPTR records stops once the first
> match is completed. All other records are discarded. From ?section 2:

Hi Alex, two comments:

1) What you say does not explain my issue (as in any case TLS or TCP
have more priority in my NAPTR's than UDP). In fact, my exact problem
is that my Kamailio is not performing NAPTR query.

2) About your quoted text, indeed RFC 2915 says that. But that just
means that, an application (in this case a SIP client) must take
*first* the valid NAPTR record with better "order" (lowest value). But
RFC 2915 is just about NAPTR, it can not mandate that applications or
protocols cannot try other NAPTR records as failover mechanism. Any
SIP client or proxy implementing NAPTR records does failover to the
next NAPTR record in case the first one failed (at least those I've
tested).

But anyhow, my main problem is that my Kamailio is not performing NAPTR query :)

Cheers.



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


From ibc at aliax.net  Thu Jun  9 19:34:29 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 9 Jun 2011 19:34:29 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=5n6c_8rKuC2yFk_N_ufoPd3WGow@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTinvGjxc=PXHRAMTb7ZowmZQubR_Fg@mail.gmail.com>
	<4DF0CA0C.5000806@gmail.com>
	<BANLkTi=5n6c_8rKuC2yFk_N_ufoPd3WGow@mail.gmail.com>
Message-ID: <BANLkTim=qWeY1O+ZbSmHXyxo5PP7cGUNYA@mail.gmail.com>

2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> According to the doc it should perform NAPTR query (it's compiled with
> USE_NAPTR as "kamailio -V" says, and parameter dns_try_naptr has value
> "yes").
>
>
> Could somebody try to send a MESSAGE or INVITE through Kamailio 3.X
> with RURI sip:whatever at oversip.net? According to NAPTR/SRV records,
> destinations should be (in order of preference):
>
> 1) TLS ?91.121.79.216:9091
> 2) TCP ?91.121.79.216:9090
> 3) UDP ?91.121.79.216:9090
>

I've tested in other Kamailio 3.X boxes, with same configuration for
DNS and support for UDP and TCP. In all cases the same, Kamailio is
*NOT* performing a NAPTR query.

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


From henning.westerholt at 1und1.de  Fri Jun 10 11:22:58 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Fri, 10 Jun 2011 11:22:58 +0200
Subject: [SR-Users] Functions for DNS
In-Reply-To: <4DF0FC3F.7010300@nothome.org>
References: <4DF0FC3F.7010300@nothome.org>
Message-ID: <201106101122.58773.henning.westerholt@1und1.de>

On Thursday 09 June 2011, Nathan Angelacos wrote:
> I'm looking to build a set of contacts with q values based on SRV
> records for serial/parallel forking.
> 
> I want "enum_query" (loading the contact set with a q value based on the
> order,preference of the NAPTR record)  but the ruri is not an e.164
> number, and SRV records are used instead.   Think in terms of "find all
> peer proxies based on SRV lookup, and sort them for use with
> t_load_contacts()/t_next_contacts()."
> 
> So far I haven't found any api that exposes the DNS lookups in sip-router.

Hi Nathan,

do you need a interface in the configuration script, or do you like to you a 
programming API to develop your own customer module?

Best regards,

Henning


From Eugen.Dedu at pu-pm.univ-fcomte.fr  Fri Jun 10 12:04:42 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Fri, 10 Jun 2011 12:04:42 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF0AA1B.9060408@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
Message-ID: <4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>

Thank you very much for your answer.

Do you confirm that for kamailio, when a user has not been online since 
a long time ago, (1) it answers with an empty body notify (and not with 
a body with Offline status), and more importantly (2) that this means 
the user is offline?

I ask this because this seems contradictory to what SIP standard says 
(see below): it does not say that empty body means offline, but just 
does not have "meaningful state".

In fact, opal (SIP library among others) developers (in CC) consider 
that empty body means no information (which is different than offline!), 
while kamailio uses it for offline!

On 09/06/11 13:10, Daniel-Constantin Mierla wrote:
> Hello,
>
> the notify is sent to inform about the state of the subscription, which
> is active in this case. If it would the first subscription to that user
> and force_active will not be set, then should be subscription state
> pending, iirc.
>
> The empty body does not change anything to the phone information about
> the watched presentity, which was known to be offline.
>
> Cheers,
> Daniel
>
> On 6/8/11 10:06 PM, Eugen Dedu wrote:
>> No idea?
>>
>> On 05/06/11 22:31, Eugen Dedu wrote:
>>> Hi,
>>>
>>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>>> users who are not registered an empty NOTIFY body is returned when asked
>>> by a SUBSCRIBE. What does this mean from SIP standard point of view, and
>>> from kamailio point of view (are they identical?) I see in
>>> RFC3265/3.1.6.2:
>>> .... If the resource
>>> has no meaningful state at the time that the SUBSCRIBE message is
>>> processed, this NOTIFY message MAY contain an empty or neutral body
>>> but is difficult for me to interpret what it means.
>>>
>>> Example: I ask the presence for a user xyz who registered and quit
>>> application long time ago:
>>>
>>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>>> CSeq: 1 SUBSCRIBE
>>> Via: SIP/2.0/UDP
>>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport
>>>
>>>
>>> User-Agent: Ekiga/3.3.1
>>> From:
>>> <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>> Supported: eventlist
>>> To: <sip:xyz at ekiga.net>
>>> Accept: application/pidf+xml
>>> Accept: multipart/related
>>> Accept: application/rlmi+xml
>>> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
>>> Allow:
>>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
>>>
>>>
>>> Expires: 300
>>> Event: presence
>>> Content-Length: 0
>>> Max-Forwards: 70
>>>
>>> I receive the following answer:
>>>
>>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>>> CSeq: 1 NOTIFY
>>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>> Contact: <sip:86.64.162.35:5060>
>>> Subscription-State: active;expires=370
>>> Event: presence
>>> Content-Length: 0
>>> Max-Forwards: 70
>>>
>>> To resume: What does SIP standard say about this NOTIFY with empty body?
>>> Does this mean that the user xyz is offline?
>>>
>>> Or does this mean that user's status has not changed? In fact, the
>>> NOTIFY with empty body (as shown above) is the first one sent by
>>> kamailio, so there is no "previous" state of that user, hence
>>> "unchanged" status has no meaning.
>>>
>>> Thank you,


From miconda at gmail.com  Fri Jun 10 12:18:45 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 12:18:45 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
Message-ID: <4DF1EF85.5070509@gmail.com>

Hello,

On 6/10/11 12:04 PM, Eugen Dedu wrote:
> Thank you very much for your answer.
>
> Do you confirm that for kamailio, when a user has not been online 
> since a long time ago, (1) it answers with an empty body notify (and 
> not with a body with Offline status), and more importantly (2) that 
> this means the user is offline?
>
> I ask this because this seems contradictory to what SIP standard says 
> (see below): it does not say that empty body means offline, but just 
> does not have "meaningful state".
>
> In fact, opal (SIP library among others) developers (in CC) consider 
> that empty body means no information (which is different than 
> offline!), while kamailio uses it for offline!
it is not for online, it is like the others understood -- an empty body 
does not change anything. Because of that I said that if the UA knew the 
user is offline, it stays offline. When a UA has a user with online 
state, if it is a notify with empty body, stays like that, nothing changes.

Cheers,
Daniel

>
> On 09/06/11 13:10, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> the notify is sent to inform about the state of the subscription, which
>> is active in this case. If it would the first subscription to that user
>> and force_active will not be set, then should be subscription state
>> pending, iirc.
>>
>> The empty body does not change anything to the phone information about
>> the watched presentity, which was known to be offline.
>>
>> Cheers,
>> Daniel
>>
>> On 6/8/11 10:06 PM, Eugen Dedu wrote:
>>> No idea?
>>>
>>> On 05/06/11 22:31, Eugen Dedu wrote:
>>>> Hi,
>>>>
>>>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>>>> users who are not registered an empty NOTIFY body is returned when 
>>>> asked
>>>> by a SUBSCRIBE. What does this mean from SIP standard point of 
>>>> view, and
>>>> from kamailio point of view (are they identical?) I see in
>>>> RFC3265/3.1.6.2:
>>>> .... If the resource
>>>> has no meaningful state at the time that the SUBSCRIBE message is
>>>> processed, this NOTIFY message MAY contain an empty or neutral body
>>>> but is difficult for me to interpret what it means.
>>>>
>>>> Example: I ask the presence for a user xyz who registered and quit
>>>> application long time ago:
>>>>
>>>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>>>> CSeq: 1 SUBSCRIBE
>>>> Via: SIP/2.0/UDP
>>>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport 
>>>>
>>>>
>>>>
>>>> User-Agent: Ekiga/3.3.1
>>>> From:
>>>> <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>> Supported: eventlist
>>>> To: <sip:xyz at ekiga.net>
>>>> Accept: application/pidf+xml
>>>> Accept: multipart/related
>>>> Accept: application/rlmi+xml
>>>> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
>>>> Allow:
>>>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK 
>>>>
>>>>
>>>>
>>>> Expires: 300
>>>> Event: presence
>>>> Content-Length: 0
>>>> Max-Forwards: 70
>>>>
>>>> I receive the following answer:
>>>>
>>>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>>>> CSeq: 1 NOTIFY
>>>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>>>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>>>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>> Contact: <sip:86.64.162.35:5060>
>>>> Subscription-State: active;expires=370
>>>> Event: presence
>>>> Content-Length: 0
>>>> Max-Forwards: 70
>>>>
>>>> To resume: What does SIP standard say about this NOTIFY with empty 
>>>> body?
>>>> Does this mean that the user xyz is offline?
>>>>
>>>> Or does this mean that user's status has not changed? In fact, the
>>>> NOTIFY with empty body (as shown above) is the first one sent by
>>>> kamailio, so there is no "previous" state of that user, hence
>>>> "unchanged" status has no meaning.
>>>>
>>>> Thank you,

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From Eugen.Dedu at pu-pm.univ-fcomte.fr  Fri Jun 10 12:27:54 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Fri, 10 Jun 2011 12:27:54 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF1EF85.5070509@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
Message-ID: <4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>

On 10/06/11 12:18, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 6/10/11 12:04 PM, Eugen Dedu wrote:
>> Thank you very much for your answer.
>>
>> Do you confirm that for kamailio, when a user has not been online
>> since a long time ago, (1) it answers with an empty body notify (and
>> not with a body with Offline status), and more importantly (2) that
>> this means the user is offline?
>>
>> I ask this because this seems contradictory to what SIP standard says
>> (see below): it does not say that empty body means offline, but just
>> does not have "meaningful state".
>>
>> In fact, opal (SIP library among others) developers (in CC) consider
>> that empty body means no information (which is different than
>> offline!), while kamailio uses it for offline!
> it is not for online, it is like the others understood -- an empty body
> does not change anything. Because of that I said that if the UA knew the
> user is offline, it stays offline. When a UA has a user with online
> state, if it is a notify with empty body, stays like that, nothing changes.

Ok.  But this happens when I start my application, so it does not know 
anything about the contacts, and it receives empty body.  So, even after 
this notify, it still does not know anything about that contact, is that 
right?

Also, as the user is offline (not connected since long time ago), why 
then kamailio sends an empty body for him instead of sending Offline 
status?  This would allow the application to correctly show him as 
offline, instead of "nothing known".  I noticed that even after several 
minutes no Offline status is sent by kamailio.

>> On 09/06/11 13:10, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> the notify is sent to inform about the state of the subscription, which
>>> is active in this case. If it would the first subscription to that user
>>> and force_active will not be set, then should be subscription state
>>> pending, iirc.
>>>
>>> The empty body does not change anything to the phone information about
>>> the watched presentity, which was known to be offline.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 6/8/11 10:06 PM, Eugen Dedu wrote:
>>>> No idea?
>>>>
>>>> On 05/06/11 22:31, Eugen Dedu wrote:
>>>>> Hi,
>>>>>
>>>>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>>>>> users who are not registered an empty NOTIFY body is returned when
>>>>> asked
>>>>> by a SUBSCRIBE. What does this mean from SIP standard point of
>>>>> view, and
>>>>> from kamailio point of view (are they identical?) I see in
>>>>> RFC3265/3.1.6.2:
>>>>> .... If the resource
>>>>> has no meaningful state at the time that the SUBSCRIBE message is
>>>>> processed, this NOTIFY message MAY contain an empty or neutral body
>>>>> but is difficult for me to interpret what it means.
>>>>>
>>>>> Example: I ask the presence for a user xyz who registered and quit
>>>>> application long time ago:
>>>>>
>>>>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>>>>> CSeq: 1 SUBSCRIBE
>>>>> Via: SIP/2.0/UDP
>>>>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport
>>>>>
>>>>>
>>>>>
>>>>> User-Agent: Ekiga/3.3.1
>>>>> From:
>>>>> <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>> Supported: eventlist
>>>>> To: <sip:xyz at ekiga.net>
>>>>> Accept: application/pidf+xml
>>>>> Accept: multipart/related
>>>>> Accept: application/rlmi+xml
>>>>> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
>>>>> Allow:
>>>>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
>>>>>
>>>>>
>>>>>
>>>>> Expires: 300
>>>>> Event: presence
>>>>> Content-Length: 0
>>>>> Max-Forwards: 70
>>>>>
>>>>> I receive the following answer:
>>>>>
>>>>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>>>>> CSeq: 1 NOTIFY
>>>>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>>>>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>>>>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>> Contact: <sip:86.64.162.35:5060>
>>>>> Subscription-State: active;expires=370
>>>>> Event: presence
>>>>> Content-Length: 0
>>>>> Max-Forwards: 70
>>>>>
>>>>> To resume: What does SIP standard say about this NOTIFY with empty
>>>>> body?
>>>>> Does this mean that the user xyz is offline?
>>>>>
>>>>> Or does this mean that user's status has not changed? In fact, the
>>>>> NOTIFY with empty body (as shown above) is the first one sent by
>>>>> kamailio, so there is no "previous" state of that user, hence
>>>>> "unchanged" status has no meaning.
>>>>>
>>>>> Thank you,
>


From nangel at nothome.org  Fri Jun 10 12:30:11 2011
From: nangel at nothome.org (Nathan Angelacos)
Date: Fri, 10 Jun 2011 06:30:11 -0400
Subject: [SR-Users] Functions for DNS
In-Reply-To: <201106101122.58773.henning.westerholt@1und1.de>
References: <4DF0FC3F.7010300@nothome.org>
	<201106101122.58773.henning.westerholt@1und1.de>
Message-ID: <4DF1F233.50205@nothome.org>

Hi Henning,

On 6/10/11 5:22 AM, Henning Westerholt wrote:
> Hi Nathan,
>
> do you need a interface in the configuration script, or do you like to you a
> programming API to develop your own customer module?
>


Interface for the configuration script.


Thanks!


From miconda at gmail.com  Fri Jun 10 12:49:02 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 12:49:02 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
Message-ID: <4DF1F69E.50000@gmail.com>



On 6/10/11 12:27 PM, Eugen Dedu wrote:
> On 10/06/11 12:18, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> On 6/10/11 12:04 PM, Eugen Dedu wrote:
>>> Thank you very much for your answer.
>>>
>>> Do you confirm that for kamailio, when a user has not been online
>>> since a long time ago, (1) it answers with an empty body notify (and
>>> not with a body with Offline status), and more importantly (2) that
>>> this means the user is offline?
>>>
>>> I ask this because this seems contradictory to what SIP standard says
>>> (see below): it does not say that empty body means offline, but just
>>> does not have "meaningful state".
>>>
>>> In fact, opal (SIP library among others) developers (in CC) consider
>>> that empty body means no information (which is different than
>>> offline!), while kamailio uses it for offline!
>> it is not for online, it is like the others understood -- an empty body
>> does not change anything. Because of that I said that if the UA knew the
>> user is offline, it stays offline. When a UA has a user with online
>> state, if it is a notify with empty body, stays like that, nothing 
>> changes.
>
> Ok.  But this happens when I start my application, so it does not know 
> anything about the contacts, and it receives empty body.  So, even 
> after this notify, it still does not know anything about that contact, 
> is that right?
Any application (or those I used so far), they show the contacts offline 
at startup, this is the initial state. Kamailio notifies with presence 
states when there is a PUBLISH by the presentity. The notification with 
the empty body give the state of the subscription, not the state of 
contact -- see the subscription-state header in notify.
>
> Also, as the user is offline (not connected since long time ago), why 
> then kamailio sends an empty body for him instead of sending Offline 
> status?  This would allow the application to correctly show him as 
> offline, instead of "nothing known".  I noticed that even after 
> several minutes no Offline status is sent by kamailio.
The sip presence server sends in notify only the presence documents 
published by a user agent. If there is no such document, the sip server 
has nothing to send. I think there are some specs that allow a user to 
set default state, probably via xcap -- but I am not sure and I haven't 
searched ietf for it.

Cheers,
Daniel

>
>>> On 09/06/11 13:10, Daniel-Constantin Mierla wrote:
>>>> Hello,
>>>>
>>>> the notify is sent to inform about the state of the subscription, 
>>>> which
>>>> is active in this case. If it would the first subscription to that 
>>>> user
>>>> and force_active will not be set, then should be subscription state
>>>> pending, iirc.
>>>>
>>>> The empty body does not change anything to the phone information about
>>>> the watched presentity, which was known to be offline.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 6/8/11 10:06 PM, Eugen Dedu wrote:
>>>>> No idea?
>>>>>
>>>>> On 05/06/11 22:31, Eugen Dedu wrote:
>>>>>> Hi,
>>>>>>
>>>>>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>>>>>> users who are not registered an empty NOTIFY body is returned when
>>>>>> asked
>>>>>> by a SUBSCRIBE. What does this mean from SIP standard point of
>>>>>> view, and
>>>>>> from kamailio point of view (are they identical?) I see in
>>>>>> RFC3265/3.1.6.2:
>>>>>> .... If the resource
>>>>>> has no meaningful state at the time that the SUBSCRIBE message is
>>>>>> processed, this NOTIFY message MAY contain an empty or neutral body
>>>>>> but is difficult for me to interpret what it means.
>>>>>>
>>>>>> Example: I ask the presence for a user xyz who registered and quit
>>>>>> application long time ago:
>>>>>>
>>>>>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>>>>>> CSeq: 1 SUBSCRIBE
>>>>>> Via: SIP/2.0/UDP
>>>>>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport 
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> User-Agent: Ekiga/3.3.1
>>>>>> From:
>>>>>> <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>>> Supported: eventlist
>>>>>> To: <sip:xyz at ekiga.net>
>>>>>> Accept: application/pidf+xml
>>>>>> Accept: multipart/related
>>>>>> Accept: application/rlmi+xml
>>>>>> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
>>>>>> Allow:
>>>>>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK 
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Expires: 300
>>>>>> Event: presence
>>>>>> Content-Length: 0
>>>>>> Max-Forwards: 70
>>>>>>
>>>>>> I receive the following answer:
>>>>>>
>>>>>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>>>>>> CSeq: 1 NOTIFY
>>>>>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>>>>>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>>>>>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>>> To: 
>>>>>> sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>>> Contact: <sip:86.64.162.35:5060>
>>>>>> Subscription-State: active;expires=370
>>>>>> Event: presence
>>>>>> Content-Length: 0
>>>>>> Max-Forwards: 70
>>>>>>
>>>>>> To resume: What does SIP standard say about this NOTIFY with empty
>>>>>> body?
>>>>>> Does this mean that the user xyz is offline?
>>>>>>
>>>>>> Or does this mean that user's status has not changed? In fact, the
>>>>>> NOTIFY with empty body (as shown above) is the first one sent by
>>>>>> kamailio, so there is no "previous" state of that user, hence
>>>>>> "unchanged" status has no meaning.
>>>>>>
>>>>>> 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



From klaus.mailinglists at pernau.at  Fri Jun 10 14:05:41 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Fri, 10 Jun 2011 14:05:41 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
Message-ID: <4DF20895.6050606@pernau.at>

My Kamailio 3.1.3 performs NATPR lookups:

use_dns_cache = yes
dns_try_naptr = yes
dns_udp_pref=1
dns_tcp_pref=1
dns_tls_pref=1
dns_sctp_pref=1

dns_use_search_list=no
dns_try_ipv6=yes
dns_retr_time=1
dns_retr_no=1


regards
Klaus


Am 09.06.2011 12:44, schrieb I?aki Baz Castillo:
> Hi, I'm testing Kamailio's NAPTR resolution:
> 
> version: kamailio 3.2.0-dev4 (x86_64/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, DBG_QM_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
> 
> 
> DNS related configuration:
> 
> dns_servers_no=2
> dns_srv_lb=yes
> dns_try_naptr=yes
> dns_use_search_list=no
> use_dns_cache=yes
> use_dns_failover=yes
> dns_tls_preference=1
> dns_tcp_preference=1
> dns_udp_preference=1
> 
> 
> As the doc says it should respect the priorities of the NAPTR records,
> see http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#dns_sctp_pref_dns_tcp_pref_dns_tls_pref_dns_udp_pref:
> 
>   "To use the remote site preferences set all dns_*_pref to the same
> positive value
>     (e.g. dns_udp_pref=1, dns_tcp_pref=1, dns_tls_pref=1, dns_sctp_pref=1)"
> 
> 
> So my kamailio receives an INVITE for an external domain oversip.net
> (no transport param neither port in the RURI) and does a t_relay.
> Request RURI is sip:qwe at oversip.net.
> 
> According to NAPTR:
> 
>   ~$ host -t naptr oversip.net
>   oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net.
>   oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net.
>   oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net.
>   oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net.
>   oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net.
> 
> So it should try TLS over TCP first, if it fails try TCP and if it
> fails try UDP.
> 
> 
> However it just uses UDP, why??
> Even if I set a minor value to dns_tls_preference (so higher priority
> I expect) it still uses UDP.
> 
> Am I doing something wrong? Thanks a lot.
> 
> 


From ibc at aliax.net  Fri Jun 10 14:23:36 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 10 Jun 2011 14:23:36 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF20895.6050606@pernau.at>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
Message-ID: <BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>

2011/6/10 Klaus Darilion <klaus.mailinglists at pernau.at>:
> My Kamailio 3.1.3 performs NATPR lookups:
>
> use_dns_cache = yes
> dns_try_naptr = yes
> dns_udp_pref=1
> dns_tcp_pref=1
> dns_tls_pref=1
> dns_sctp_pref=1
>
> dns_use_search_list=no
> dns_try_ipv6=yes
> dns_retr_time=1
> dns_retr_no=1


With your exact configuration, my kamailio 3.2.0-dev5 (debian package)
does not perform NAPTR :(

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


From ibc at aliax.net  Fri Jun 10 14:30:03 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 10 Jun 2011 14:30:03 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
Message-ID: <BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>

2011/6/10 I?aki Baz Castillo <ibc at aliax.net>:
> With your exact configuration, my kamailio 3.2.0-dev5 (debian package)
> does not perform NAPTR :(

Klaus, could you provide me the following information?


1) kamailio -V

2) Is compiled? a deb package?

3) Could you please call sip:anything at oversip.net and show me kamailio
logs by filtering "dns" ( grep | dns ) ?


Really thanks a lot.

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


From klaus.mailinglists at pernau.at  Fri Jun 10 15:14:17 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Fri, 10 Jun 2011 15:14:17 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>	<4DF20895.6050606@pernau.at>	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
Message-ID: <4DF218A9.7050205@pernau.at>

Kamailio performs NAPTR lookups, but chooses UDP. I do not know why.


# kamailio -V
version: kamailio 3.1.3 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
USE_STUN, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP,
PKG_MALLOC, DBG_QM_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, PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 14:01:24 May 13 2011 with gcc 4.3.2


# tail -f /var/log/kamailio|grep dns

<core> [dns_cache.c:567]: dns_hash_find(oversip.net(11), 35), h=956
<core> [dns_cache.c:1777]: dns_get_related(0xb3921838 (oversip.net, 35),
35, *0x8366a5c) (0)
<core> [dns_cache.c:870]: dns_cache_add: adding oversip.net(11) 35
(flags=0) at 956
<core> [dns_cache.c:567]: dns_hash_find(_sip._udp.oversip.net(21), 33), h=23
<core> [dns_cache.c:1777]: dns_get_related(0xb3921a08
(_sip._udp.oversip.net, 33), 33, *0x8366a5c) (0)
<core> [dns_cache.c:870]: dns_cache_add: adding
_sip._udp.oversip.net(21) 33 (flags=0) at 23
<core> [dns_cache.c:870]: dns_cache_add: adding sip.oversip.net(15) 1
(flags=0) at 400
<core> [dns_cache.c:567]: dns_hash_find(sip.oversip.net(15), 1), h=400


# tshark host 83.136.32.159 or port 53
A -> B TCP 4923 > sip [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=3
B -> A TCP sip > 4923 [SYN, ACK] Seq=0 Ack=1 Win=5840 Len=0 MSS=1460 WS=6
A -> B TCP 4923 > sip [ACK] Seq=1 Ack=1 Win=415024 Len=0
A -> B SIP/SDP Request: INVITE sip:anything at oversip.net, with session
description
B -> A TCP sip > 4923 [ACK] Seq=1 Ack=867 Win=8768 Len=0
B -> A SIP Status: 407 Proxy Authentication Required
A -> B SIP Request: ACK sip:anything at oversip.net
B -> A TCP sip > 4923 [ACK] Seq=676 Ack=1239 Win=10496 Len=0
A -> B SIP/SDP Request: INVITE sip:anything at oversip.net, with session
description
B -> A TCP sip > 4923 [ACK] Seq=676 Ack=2310 Win=13440 Len=0
B -> A SIP Status: 100 your wish is my command,
ru=sip:anything at oversip.net, du=<null>
C -> D DNS Standard query NAPTR oversip.net
D -> C DNS Standard query response NAPTR 20 50 S NAPTR 40 50 S NAPTR 50
50 S NAPTR 5 50 S NAPTR 10 50 S
C -> D DNS Standard query SRV _sip._udp.oversip.net
D -> C DNS Standard query response SRV 0 0 9090 sip.oversip.net
B -> D SIP/SDP Request: INVITE sip:anything at oversip.net, with session
description
D -> B SIP Status: 555 Ok, I'm here, your request arrives via UDP
B -> D SIP Request: ACK sip:anything at oversip.net
B -> A SIP Status: 555 Ok, I'm here, your request arrives via UDP
A -> B TCP 4923 > sip [ACK] Seq=2310 Ack=1682 Win=415024 Len=0
A -> B SIP Request: ACK sip:anything at oversip.net
B -> A TCP sip > 4923 [ACK] Seq=1682 Ack=2662 Win=15616 Len=0











Am 10.06.2011 14:30, schrieb I?aki Baz Castillo:
> 2011/6/10 I?aki Baz Castillo <ibc at aliax.net>:
>> With your exact configuration, my kamailio 3.2.0-dev5 (debian package)
>> does not perform NAPTR :(
> 
> Klaus, could you provide me the following information?
> 
> 
> 1) kamailio -V
> 
> 2) Is compiled? a deb package?
> 
> 3) Could you please call sip:anything at oversip.net and show me kamailio
> logs by filtering "dns" ( grep | dns ) ?
> 
> 
> Really thanks a lot.
> 


From ibc at aliax.net  Fri Jun 10 15:33:34 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 10 Jun 2011 15:33:34 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF218A9.7050205@pernau.at>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
Message-ID: <BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>

2011/6/10 Klaus Darilion <klaus.mailinglists at pernau.at>:
> Kamailio performs NAPTR lookups, but chooses UDP. I do not know why.

You are fully right. I've captured DNS traffic and wireshark clearly
shows a NAPTR record. And indeed kamailio prefers SIP over UDP with no
reason (it should prefer TCP as my kamailio does not speak TLS):

$ host -t naptr oversip.net
oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net.
oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net.
oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net.
oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net.
oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net.


Maybe kamailio is taking the "order" field of the NAPTR incorrectly
and gives priority with higher values? I will check it ASAP.

Thanks a lot Klaus.

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


From miconda at gmail.com  Fri Jun 10 16:07:21 2011
From: miconda at gmail.com (Daniel-Constantine Mierla)
Date: Fri, 10 Jun 2011 16:07:21 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
Message-ID: <3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>



On Jun 10, 2011, at 12:27 PM, Eugen Dedu <Eugen.Dedu at pu-pm.univ-fcomte.fr> wrote:

> On 10/06/11 12:18, Daniel-Constantin Mierla wrote:
>> Hello,
>> 
>> On 6/10/11 12:04 PM, Eugen Dedu wrote:
>>> Thank you very much for your answer.
>>> 
>>> Do you confirm that for kamailio, when a user has not been online
>>> since a long time ago, (1) it answers with an empty body notify (and
>>> not with a body with Offline status), and more importantly (2) that
>>> this means the user is offline?
>>> 
>>> I ask this because this seems contradictory to what SIP standard says
>>> (see below): it does not say that empty body means offline, but just
>>> does not have "meaningful state".
>>> 
>>> In fact, opal (SIP library among others) developers (in CC) consider
>>> that empty body means no information (which is different than
>>> offline!), while kamailio uses it for offline!
>> it is not for online, it is like the others understood -- an empty body
>> does not change anything. Because of that I said that if the UA knew the
>> user is offline, it stays offline. When a UA has a user with online
>> state, if it is a notify with empty body, stays like that, nothing changes.
> 
> Ok.  But this happens when I start my application, so it does not know anything about the contacts, and it receives empty body.  So, even after this notify, it still does not know anything about that contact, is that right?

What happens if there is no reply to subscribe and no notify? Is it considered offline? If yes, then probably should stay the same for first notify with empty body.

Iirc it is a must to send a body to tell the subscription state.

> 
> Also, as the user is offline (not connected since long time ago), why then kamailio sends an empty body for him instead of sending Offline status?  This would allow the application to correctly show him as offline, instead of "nothing known".  I noticed that even after several minutes no Offline status is sent by kamailio.

The presence server sends notifications with documents published by phones. If there is no such document, I see no reason to build one, unless the user demanded that via some preferences - it would be xcap - I haven't searched the specs, just my expectation based on simple presence architecture.

> 
>>> On 09/06/11 13:10, Daniel-Constantin Mierla wrote:
>>>> Hello,
>>>> 
>>>> the notify is sent to inform about the state of the subscription, which
>>>> is active in this case. If it would the first subscription to that user
>>>> and force_active will not be set, then should be subscription state
>>>> pending, iirc.
>>>> 
>>>> The empty body does not change anything to the phone information about
>>>> the watched presentity, which was known to be offline.
>>>> 
>>>> Cheers,
>>>> Daniel
>>>> 
>>>> On 6/8/11 10:06 PM, Eugen Dedu wrote:
>>>>> No idea?
>>>>> 
>>>>> On 05/06/11 22:31, Eugen Dedu wrote:
>>>>>> Hi,
>>>>>> 
>>>>>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>>>>>> users who are not registered an empty NOTIFY body is returned when
>>>>>> asked
>>>>>> by a SUBSCRIBE. What does this mean from SIP standard point of
>>>>>> view, and
>>>>>> from kamailio point of view (are they identical?) I see in
>>>>>> RFC3265/3.1.6.2:
>>>>>> .... If the resource
>>>>>> has no meaningful state at the time that the SUBSCRIBE message is
>>>>>> processed, this NOTIFY message MAY contain an empty or neutral body
>>>>>> but is difficult for me to interpret what it means.
>>>>>> 
>>>>>> Example: I ask the presence for a user xyz who registered and quit
>>>>>> application long time ago:
>>>>>> 
>>>>>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>>>>>> CSeq: 1 SUBSCRIBE
>>>>>> Via: SIP/2.0/UDP
>>>>>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> User-Agent: Ekiga/3.3.1
>>>>>> From:
>>>>>> <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>>> Supported: eventlist
>>>>>> To: <sip:xyz at ekiga.net>
>>>>>> Accept: application/pidf+xml
>>>>>> Accept: multipart/related
>>>>>> Accept: application/rlmi+xml
>>>>>> Contact: <sip:eugen.dedu at 82.238.108.175:5060>
>>>>>> Allow:
>>>>>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Expires: 300
>>>>>> Event: presence
>>>>>> Content-Length: 0
>>>>>> Max-Forwards: 70
>>>>>> 
>>>>>> I receive the following answer:
>>>>>> 
>>>>>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>>>>>> CSeq: 1 NOTIFY
>>>>>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>>>>>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>>>>>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>>> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>>> Contact: <sip:86.64.162.35:5060>
>>>>>> Subscription-State: active;expires=370
>>>>>> Event: presence
>>>>>> Content-Length: 0
>>>>>> Max-Forwards: 70
>>>>>> 
>>>>>> To resume: What does SIP standard say about this NOTIFY with empty
>>>>>> body?
>>>>>> Does this mean that the user xyz is offline?
>>>>>> 
>>>>>> Or does this mean that user's status has not changed? In fact, the
>>>>>> NOTIFY with empty body (as shown above) is the first one sent by
>>>>>> kamailio, so there is no "previous" state of that user, hence
>>>>>> "unchanged" status has no meaning.
>>>>>> 
>>>>>> Thank you,
>> 


From spinov_evgeniy at intalisan.com  Fri Jun 10 18:43:22 2011
From: spinov_evgeniy at intalisan.com (Evgeniy Spinov)
Date: Fri, 10 Jun 2011 17:43:22 +0100
Subject: [SR-Users] Kamailio dispatcher
Message-ID: <1307724203.1565.11.camel@localhost.localdomain>

Hello.

I had a Kamailio version of 3.0.3 and during this time configured a
failover with simple routine. In short like this:

if (ds_next_domain()) {
	xlog(.....);
	if (!t_relay()) {
		xlog(.....);
		return;
	}
	return;
} else {
	t_reply("503");
}

Then I've updated to 3.1.3 and was happy enough until I've discovered
that my failover is not working anymore. 

What is observed: 
- It changes $rd variable, but not changing $du variable, is it ok?
- Not any of the nodes receive packet on t_relay after ds_next_domain().
- In route decision section, where t_on_reply() and t_on_failure() are
located I have the same t_relay() and it works fine, cause there is no
any ds_next_domains().

Is it a bug or I'm doing something wrong?

Thanks in advance.




From Eugen.Dedu at pu-pm.univ-fcomte.fr  Fri Jun 10 19:58:04 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Fri, 10 Jun 2011 19:58:04 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
Message-ID: <4DF25B2C.4060906@pu-pm.univ-fcomte.fr>

On 10/06/11 16:07, Daniel-Constantine Mierla wrote:
>
>
> On Jun 10, 2011, at 12:27 PM, Eugen Dedu<Eugen.Dedu at pu-pm.univ-fcomte.fr>  wrote:
>
>> On 10/06/11 12:18, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> On 6/10/11 12:04 PM, Eugen Dedu wrote:
>>>> Thank you very much for your answer.
>>>>
>>>> Do you confirm that for kamailio, when a user has not been online
>>>> since a long time ago, (1) it answers with an empty body notify (and
>>>> not with a body with Offline status), and more importantly (2) that
>>>> this means the user is offline?
>>>>
>>>> I ask this because this seems contradictory to what SIP standard says
>>>> (see below): it does not say that empty body means offline, but just
>>>> does not have "meaningful state".
>>>>
>>>> In fact, opal (SIP library among others) developers (in CC) consider
>>>> that empty body means no information (which is different than
>>>> offline!), while kamailio uses it for offline!
>>> it is not for online, it is like the others understood -- an empty body
>>> does not change anything. Because of that I said that if the UA knew the
>>> user is offline, it stays offline. When a UA has a user with online
>>> state, if it is a notify with empty body, stays like that, nothing changes.
>>
>> Ok.  But this happens when I start my application, so it does not know anything about the contacts, and it receives empty body.  So, even after this notify, it still does not know anything about that contact, is that right?
>
> What happens if there is no reply to subscribe and no notify? Is it
considered offline? If yes, then probably should stay the same for first
notify with empty body.

If the application starts by putting each contact as offline, there will 
be no difference between no information known (while waiting a notify) 
and offline.

Moreover, no reply to subscribe could be another status: presence error.

So I would say the right thing is to have three statuses:
- no info known
- presence error
- offline

To make things simple, I will put everyone in my application as offline 
at the beginning, and the user will have to wait a few seconds (how 
many?) to discover if some contact is really offline or still waiting a 
notify.

Thank you so far for your explanations.

> Iirc it is a must to send a body to tell the subscription state.
>
>>
>> Also, as the user is offline (not connected since long time ago), why then kamailio sends an empty body for him instead of sending Offline status?  This would allow the application to correctly show him as offline, instead of "nothing known".  I noticed that even after several minutes no Offline status is sent by kamailio.
>
> The presence server sends notifications with documents published by phones. If there is no such document, I see no reason to build one, unless the user demanded that via some preferences - it would be xcap - I haven't searched the specs, just my expectation based on simple presence architecture.
>
>>
>>>> On 09/06/11 13:10, Daniel-Constantin Mierla wrote:
>>>>> Hello,
>>>>>
>>>>> the notify is sent to inform about the state of the subscription, which
>>>>> is active in this case. If it would the first subscription to that user
>>>>> and force_active will not be set, then should be subscription state
>>>>> pending, iirc.
>>>>>
>>>>> The empty body does not change anything to the phone information about
>>>>> the watched presentity, which was known to be offline.
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> On 6/8/11 10:06 PM, Eugen Dedu wrote:
>>>>>> No idea?
>>>>>>
>>>>>> On 05/06/11 22:31, Eugen Dedu wrote:
>>>>>>> Hi,
>>>>>>>
>>>>>>> ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for
>>>>>>> users who are not registered an empty NOTIFY body is returned when
>>>>>>> asked
>>>>>>> by a SUBSCRIBE. What does this mean from SIP standard point of
>>>>>>> view, and
>>>>>>> from kamailio point of view (are they identical?) I see in
>>>>>>> RFC3265/3.1.6.2:
>>>>>>> .... If the resource
>>>>>>> has no meaningful state at the time that the SUBSCRIBE message is
>>>>>>> processed, this NOTIFY message MAY contain an empty or neutral body
>>>>>>> but is difficult for me to interpret what it means.
>>>>>>>
>>>>>>> Example: I ask the presence for a user xyz who registered and quit
>>>>>>> application long time ago:
>>>>>>>
>>>>>>> SUBSCRIBE sip:xyz at ekiga.net SIP/2.0
>>>>>>> CSeq: 1 SUBSCRIBE
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> User-Agent: Ekiga/3.3.1
>>>>>>> From:
>>>>>>> <sip:eugen.dedu at ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>>>> Supported: eventlist
>>>>>>> To:<sip:xyz at ekiga.net>
>>>>>>> Accept: application/pidf+xml
>>>>>>> Accept: multipart/related
>>>>>>> Accept: application/rlmi+xml
>>>>>>> Contact:<sip:eugen.dedu at 82.238.108.175:5060>
>>>>>>> Allow:
>>>>>>> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Expires: 300
>>>>>>> Event: presence
>>>>>>> Content-Length: 0
>>>>>>> Max-Forwards: 70
>>>>>>>
>>>>>>> I receive the following answer:
>>>>>>>
>>>>>>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>>>>>>> CSeq: 1 NOTIFY
>>>>>>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0
>>>>>>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>>>>>>> From: sip:xyz at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0
>>>>>>> Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8 at snoopy
>>>>>>> To: sip:eugen.dedu at ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8
>>>>>>> Contact:<sip:86.64.162.35:5060>
>>>>>>> Subscription-State: active;expires=370
>>>>>>> Event: presence
>>>>>>> Content-Length: 0
>>>>>>> Max-Forwards: 70
>>>>>>>
>>>>>>> To resume: What does SIP standard say about this NOTIFY with empty
>>>>>>> body?
>>>>>>> Does this mean that the user xyz is offline?
>>>>>>>
>>>>>>> Or does this mean that user's status has not changed? In fact, the
>>>>>>> NOTIFY with empty body (as shown above) is the first one sent by
>>>>>>> kamailio, so there is no "previous" state of that user, hence
>>>>>>> "unchanged" status has no meaning.
>>>>>>>
>>>>>>> Thank you,
>>>
>


From miconda at gmail.com  Fri Jun 10 20:08:09 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 20:08:09 +0200
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <1307724203.1565.11.camel@localhost.localdomain>
References: <1307724203.1565.11.camel@localhost.localdomain>
Message-ID: <4DF25D89.9090104@gmail.com>

Hello,

to understand the scenario:
- first branch has destination uri ($du) set
- it failed and gets to failure route where you call ds_next_domain() 
and $du s still the one from first branch?

What do you mean that "not any of the nodes receive the packet ..."?

Cheers.
Daniel

On 6/10/11 6:43 PM, Evgeniy Spinov wrote:
> Hello.
>
> I had a Kamailio version of 3.0.3 and during this time configured a
> failover with simple routine. In short like this:
>
> if (ds_next_domain()) {
> 	xlog(.....);
> 	if (!t_relay()) {
> 		xlog(.....);
> 		return;
> 	}
> 	return;
> } else {
> 	t_reply("503");
> }
>
> Then I've updated to 3.1.3 and was happy enough until I've discovered
> that my failover is not working anymore.
>
> What is observed:
> - It changes $rd variable, but not changing $du variable, is it ok?
> - Not any of the nodes receive packet on t_relay after ds_next_domain().
> - In route decision section, where t_on_reply() and t_on_failure() are
> located I have the same t_relay() and it works fine, cause there is no
> any ds_next_domains().
>
> Is it a bug or I'm doing something wrong?
>
> 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



From miconda at gmail.com  Fri Jun 10 20:20:35 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 20:20:35 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF25B2C.4060906@pu-pm.univ-fcomte.fr>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr>
Message-ID: <4DF26073.30001@gmail.com>



On 6/10/11 7:58 PM, Eugen Dedu wrote:
> [...]
>
> If the application starts by putting each contact as offline, there 
> will be no difference between no information known (while waiting a 
> notify) and offline.
>
> Moreover, no reply to subscribe could be another status: presence error.
>
> So I would say the right thing is to have three statuses:
> - no info known
> - presence error
> - offline
>
> To make things simple, I will put everyone in my application as 
> offline at the beginning, and the user will have to wait a few seconds 
> (how many?) to discover if some contact is really offline or still 
> waiting a notify.
>
> Thank you so far for your explanations.
have you spotted in SIP/SIMPLE specs what has to be sent when the 
presentity is offline? My quick google was not that succesful. Maybe 
Inaki has the specs more fresh indexed in memory and can help.

I based my assumption that the sip/simple presence server should not 
generate itself a presence/pidf document with status closed/offline 
since the pidf has a <tuple id="..."> node that is generated by 
presentity device.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Fri Jun 10 20:23:17 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 10 Jun 2011 20:23:17 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
Message-ID: <BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>

2011/6/10 I?aki Baz Castillo <ibc at aliax.net>:
> Maybe kamailio is taking the "order" field of the NAPTR incorrectly
> and gives priority with higher values? I will check it ASAP.

Ok, I've created domain whose NAPTR  record just includes a single
SIP+D2T entry (just SIP over TCP).

My Kamailio (which uses UDP and TCP) indeed performs the NAPTR
queries, but it seems it doesn't like the response and ignores it.
Then it performs an usual _sip._udp SRP query.

My DNS domains and NAPTR/SRV/A records work OK with any other SIP
client implementing RFC 3263.

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


From miconda at gmail.com  Fri Jun 10 21:58:59 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 21:58:59 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF25FD4.8080107@southeren.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<4DF1F69E.50000@gmail.com> <4DF25FD4.8080107@southeren.com>
Message-ID: <4DF27783.8080903@gmail.com>



On 6/10/11 8:17 PM, Craig Southeren wrote:
> On 10/06/2011 3:49 AM, Daniel-Constantin Mierla wrote:
>
> ..deleted
>>> Ok.  But this happens when I start my application, so it does not 
>>> know anything about the contacts, and it receives empty body.  So, 
>>> even after this notify, it still does not know anything about that 
>>> contact, is that right?
>>
>> Any application (or those I used so far), they show the contacts 
>> offline at startup, this is the initial state. Kamailio notifies with 
>> presence states when there is a PUBLISH by the presentity. The 
>> notification with the empty body give the state of the subscription, 
>> not the state of contact -- see the subscription-state header in notify.
>>>
>>> Also, as the user is offline (not connected since long time ago), 
>>> why then kamailio sends an empty body for him instead of sending 
>>> Offline status?  This would allow the application to correctly show 
>>> him as offline, instead of "nothing known".  I noticed that even 
>>> after several minutes no Offline status is sent by kamailio.
>> The sip presence server sends in notify only the presence documents 
>> published by a user agent. If there is no such document, the sip 
>> server has nothing to send. I think there are some specs that allow a 
>> user to set default state, probably via xcap -- but I am not sure and 
>> I haven't searched ietf for it.
>
> I agree that an application should assume a contact is offline at the 
> time of the initial SUBSCRIBE. Any subsequent NOTIFYs will modify that 
> status.
>
> If an empty NOTIFY is send as the first NOTIFY after a SUBSCRIBE, then 
> this means the server has no presence document for the subscribed 
> entity.  This will occur when when entity is offline.
>
> If the entity is online, then the first NOTIFY may still be empty, but 
> there should be a subsequent NOTIFY containing the current presence 
> document.
>
> Hopefully, this second NOTIFY is sent as soon as possible after the 
> SUBSCRIBE, and is not deferrred to when the second party refreshes 
> it's presence document, which could be minutes or even hours later 
> (depending on the presence timeout)
>
> Daniel - can you confirn this?
yes, this is how I thought it should be.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Fri Jun 10 22:01:01 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 22:01:01 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF2672F.2080006@southeren.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr>
	<4DF26073.30001@gmail.com> <4DF2672F.2080006@southeren.com>
Message-ID: <4DF277FD.8050908@gmail.com>



On 6/10/11 8:49 PM, Craig Southeren wrote:
> On 10/06/2011 11:20 AM, Daniel-Constantin Mierla wrote:
>
> ...deleted
>>> Thank you so far for your explanations.
>>
>> have you spotted in SIP/SIMPLE specs what has to be sent when the 
>> presentity is offline? My quick google was not that succesful. Maybe 
>> Inaki has the specs more fresh indexed in memory and can help.
>>
>> I based my assumption that the sip/simple presence server should not 
>> generate itself a presence/pidf document with status closed/offline 
>> since the pidf has a <tuple id="..."> node that is generated by 
>> presentity device.
>
> I know there is an XCAP verb to do this.
>
> Also, you might like to look at "RFC 4481 - Timed Presence Extensions 
> to the Presence Information Data Format". This allows a client to 
> specify status for a future time period when other status may not be 
> available.
>
> Specifically, the RFC says:
>
>   "During composition, a presence agent (PA) may encounter a stored 
> <timed-status> element that covers the present time.  The PA MAY 
> either discard that element or MAY
>    convert it to a regular <status> element if it considers that 
> information more credible."
>
> So, if a client uploads a presence document that contains a 
> <timed-status> element, that element could be sent by the PA when the 
> client goes offline, rather than the entire document being discarded 
> (which is what I suspect is happening now).
we don't have this rfc implemented in kamailio at this time. Is any UA 
you know doing this kind of thing?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Fri Jun 10 22:06:15 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 22:06:15 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
Message-ID: <4DF27937.8000803@gmail.com>



On 6/10/11 8:23 PM, I?aki Baz Castillo wrote:
> 2011/6/10 I?aki Baz Castillo<ibc at aliax.net>:
>> Maybe kamailio is taking the "order" field of the NAPTR incorrectly
>> and gives priority with higher values? I will check it ASAP.
> Ok, I've created domain whose NAPTR  record just includes a single
> SIP+D2T entry (just SIP over TCP).
>
> My Kamailio (which uses UDP and TCP) indeed performs the NAPTR
> queries, but it seems it doesn't like the response and ignores it.
> Then it performs an usual _sip._udp SRP query.
>
> My DNS domains and NAPTR/SRV/A records work OK with any other SIP
> client implementing RFC 3263.
>
Could you set children to 1, attach with gdb to the sip worker handling 
the invite, send the call and execute step by step to see which 
condition fails? A simple config with t_relay() should make the 
debugging easier.

Thanks,
Daniel

-- 
Daniel-Constantin Mierla
http://www.asipto.com



From craig at southeren.com  Fri Jun 10 20:17:56 2011
From: craig at southeren.com (Craig Southeren)
Date: Fri, 10 Jun 2011 11:17:56 -0700
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF1F69E.50000@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr> <4DF1F69E.50000@gmail.com>
Message-ID: <4DF25FD4.8080107@southeren.com>

On 10/06/2011 3:49 AM, Daniel-Constantin Mierla wrote:

..deleted
>> Ok.  But this happens when I start my application, so it does not 
>> know anything about the contacts, and it receives empty body.  So, 
>> even after this notify, it still does not know anything about that 
>> contact, is that right?
>
> Any application (or those I used so far), they show the contacts 
> offline at startup, this is the initial state. Kamailio notifies with 
> presence states when there is a PUBLISH by the presentity. The 
> notification with the empty body give the state of the subscription, 
> not the state of contact -- see the subscription-state header in notify.
>>
>> Also, as the user is offline (not connected since long time ago), why 
>> then kamailio sends an empty body for him instead of sending Offline 
>> status?  This would allow the application to correctly show him as 
>> offline, instead of "nothing known".  I noticed that even after 
>> several minutes no Offline status is sent by kamailio.
> The sip presence server sends in notify only the presence documents 
> published by a user agent. If there is no such document, the sip 
> server has nothing to send. I think there are some specs that allow a 
> user to set default state, probably via xcap -- but I am not sure and 
> I haven't searched ietf for it.

I agree that an application should assume a contact is offline at the 
time of the initial SUBSCRIBE. Any subsequent NOTIFYs will modify that 
status.

If an empty NOTIFY is send as the first NOTIFY after a SUBSCRIBE, then 
this means the server has no presence document for the subscribed 
entity.  This will occur when when entity is offline.

If the entity is online, then the first NOTIFY may still be empty, but 
there should be a subsequent NOTIFY containing the current presence 
document.

Hopefully, this second NOTIFY is sent as soon as possible after the 
SUBSCRIBE, and is not deferrred to when the second party refreshes it's 
presence document, which could be minutes or even hours later (depending 
on the presence timeout)

Daniel - can you confirn this?

    Craig

--

-----------------------------------------------------------------------
  Craig Southeren          Post Increment ? VoIP Consulting and Software
  craigs at postincrement.com.au                   www.postincrement.com.au

  Phone:  +61 243654666      ICQ: #86852844
  Fax:    +61 243656905      MSN: craig_southeren at hotmail.com
  Mobile: +61 417231046      Jabber: craigs at jabber.org

  "Science is the poetry of reality." Richard Dawkins



From craig at southeren.com  Fri Jun 10 20:49:19 2011
From: craig at southeren.com (Craig Southeren)
Date: Fri, 10 Jun 2011 11:49:19 -0700
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF26073.30001@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr> <4DF26073.30001@gmail.com>
Message-ID: <4DF2672F.2080006@southeren.com>

On 10/06/2011 11:20 AM, Daniel-Constantin Mierla wrote:

...deleted
>> Thank you so far for your explanations.
>
> have you spotted in SIP/SIMPLE specs what has to be sent when the 
> presentity is offline? My quick google was not that succesful. Maybe 
> Inaki has the specs more fresh indexed in memory and can help.
>
> I based my assumption that the sip/simple presence server should not 
> generate itself a presence/pidf document with status closed/offline 
> since the pidf has a <tuple id="..."> node that is generated by 
> presentity device.

I know there is an XCAP verb to do this.

Also, you might like to look at "RFC 4481 - Timed Presence Extensions to 
the Presence Information Data Format". This allows a client to specify 
status for a future time period when other status may not be available.

Specifically, the RFC says:

   "During composition, a presence agent (PA) may encounter a stored 
<timed-status> element that covers the present time.  The PA MAY either 
discard that element or MAY
    convert it to a regular <status> element if it considers that 
information more credible."

So, if a client uploads a presence document that contains a 
<timed-status> element, that element could be sent by the PA when the 
client goes offline, rather than the entire document being discarded 
(which is what I suspect is happening now).

    Craig

-- 

-----------------------------------------------------------------------
  Craig Southeren          Post Increment ? VoIP Consulting and Software
  craigs at postincrement.com.au                   www.postincrement.com.au

  Phone:  +61 243654666      ICQ: #86852844
  Fax:    +61 243656905      MSN: craig_southeren at hotmail.com
  Mobile: +61 417231046      Jabber: craigs at jabber.org

  "Science is the poetry of reality." Richard Dawkins



From Eugen.Dedu at pu-pm.univ-fcomte.fr  Fri Jun 10 22:17:06 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Fri, 10 Jun 2011 22:17:06 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF27783.8080903@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<4DF1F69E.50000@gmail.com> <4DF25FD4.8080107@southeren.com>
	<4DF27783.8080903@gmail.com>
Message-ID: <4DF27BC2.3030206@pu-pm.univ-fcomte.fr>

On 10/06/11 21:58, Daniel-Constantin Mierla wrote:
>
>
> On 6/10/11 8:17 PM, Craig Southeren wrote:
>> On 10/06/2011 3:49 AM, Daniel-Constantin Mierla wrote:
>>
>> ..deleted
>>>> Ok. But this happens when I start my application, so it does not
>>>> know anything about the contacts, and it receives empty body. So,
>>>> even after this notify, it still does not know anything about that
>>>> contact, is that right?
>>>
>>> Any application (or those I used so far), they show the contacts
>>> offline at startup, this is the initial state. Kamailio notifies with
>>> presence states when there is a PUBLISH by the presentity. The
>>> notification with the empty body give the state of the subscription,
>>> not the state of contact -- see the subscription-state header in notify.
>>>>
>>>> Also, as the user is offline (not connected since long time ago),
>>>> why then kamailio sends an empty body for him instead of sending
>>>> Offline status? This would allow the application to correctly show
>>>> him as offline, instead of "nothing known". I noticed that even
>>>> after several minutes no Offline status is sent by kamailio.
>>> The sip presence server sends in notify only the presence documents
>>> published by a user agent. If there is no such document, the sip
>>> server has nothing to send. I think there are some specs that allow a
>>> user to set default state, probably via xcap -- but I am not sure and
>>> I haven't searched ietf for it.
>>
>> I agree that an application should assume a contact is offline at the
>> time of the initial SUBSCRIBE. Any subsequent NOTIFYs will modify that
>> status.
>>
>> If an empty NOTIFY is send as the first NOTIFY after a SUBSCRIBE, then
>> this means the server has no presence document for the subscribed
>> entity. This will occur when when entity is offline.
>>
>> If the entity is online, then the first NOTIFY may still be empty, but
>> there should be a subsequent NOTIFY containing the current presence
>> document.
>>
>> Hopefully, this second NOTIFY is sent as soon as possible after the
>> SUBSCRIBE, and is not deferrred to when the second party refreshes
>> it's presence document, which could be minutes or even hours later
>> (depending on the presence timeout)
>>
>> Daniel - can you confirn this?
> yes, this is how I thought it should be.

It seems logical to me too.

Thank you, Daniel,
-- 
Eugen


From miconda at gmail.com  Fri Jun 10 22:22:31 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 22:22:31 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF27981.1040706@southeren.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr>
	<4DF26073.30001@gmail.com> <4DF2672F.2080006@southeren.com>
	<4DF277FD.8050908@gmail.com> <4DF27981.1040706@southeren.com>
Message-ID: <4DF27D07.9020901@gmail.com>



On 6/10/11 10:07 PM, Craig Southeren wrote:
> On 10/06/2011 1:01 PM, Daniel-Constantin Mierla wrote:..deleted
>>>
>>> So, if a client uploads a presence document that contains a 
>>> <timed-status> element, that element could be sent by the PA when 
>>> the client goes offline, rather than the entire document being 
>>> discarded (which is what I suspect is happening now).
>>
>> we don't have this rfc implemented in kamailio at this time. Is any 
>> UA you know doing this kind of thing?
>
> No
Then it is the time for Eugen to add it in his sip client and send the 
patch for enhancing kamailio's presence server with rfc4481, as well  ;-)

Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From Eugen.Dedu at pu-pm.univ-fcomte.fr  Fri Jun 10 22:32:27 2011
From: Eugen.Dedu at pu-pm.univ-fcomte.fr (Eugen Dedu)
Date: Fri, 10 Jun 2011 22:32:27 +0200
Subject: [SR-Users] Double tuple in notify
Message-ID: <4DF27F5B.5020501@pu-pm.univ-fcomte.fr>

Hi again,

> Then it is the time for Eugen to add it in his sip client and send
> the patch for enhancing kamailio's presence server with rfc4481, as
> well

:o)

I start my application, and make my own status be shown.  I change my 
status back and forth a few times between Away and DoNotDisturb (each 
time a publish is sent), the status returned (in notify) is good.  When 
I change my status to Online/Available (and only to this), kamailio 
returns a double tuple.  Is it normal to have a double tuple?  If yes, 
which one to choose (the one with latest timestamp)?  Here it is:

NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
CSeq: 7 NOTIFY
Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK7634.4d863c14.0
User-Agent: Kamailio (1.5.3-notls (i386/linux))
From: sip:eugen.dedu at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-29fc
Call-ID: 8a0d723b-0f8d-e011-84d7-0024d693d8e8 at snoopy
To: sip:eugen.dedu at ekiga.net;tag=7c75723b-0f8d-e011-84d7-0024d693d8e8
Contact: <sip:86.64.162.35:5060>
Subscription-State: active;expires=270
Event: presence
Content-Length: 938
Content-Type: application/pidf+xml
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" 
xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:
rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:eugen.dedu at ekiga.net">
   <tuple id="T00000002">
     <status>
       <basic>open</basic>
     </status>
     <contact priority="1">sip:eugen.dedu at ekiga.net</contact>
     <note>online - </note>
     <timestamp>2011-06-04T13:55:56+02:00+02:00</timestamp>
   </tuple>

   <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="T00000001">
     <status>
       <basic>open</basic>
     </status>
     <contact priority="1">sip:eugen.dedu at ekiga.net</contact>
     <note>away - jjk</note>
     <timestamp>2011-06-04T13:54:37+02:00+02:00</timestamp>
   </tuple>
   <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" 
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="pid1">
     <rpid:activities>
       <rpid:away/>
     </rpid:activities>
   </dm:person>
</presence>

-- 
Eugen


From ibc at aliax.net  Fri Jun 10 23:16:16 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 10 Jun 2011 23:16:16 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF27937.8000803@gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
Message-ID: <BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>

2011/6/10 Daniel-Constantin Mierla <miconda at gmail.com>:
> Could you set children to 1, attach with gdb to the sip worker handling the
> invite, send the call and execute step by step to see which condition fails?
> A simple config with t_relay() should make the debugging easier.

Hi Daniel. Which worker should I attach? My kamailio listens in UDP
and TCP. I send the request via UDP and it should do NAPTR stuf and
choose SIP TCP (best priority). So:

  Process::  ID=0 PID=26913 Type=attendant
  Process::  ID=1 PID=26914 Type=udp receiver child=0 sock=192.168.1.16:8080
  Process::  ID=2 PID=26915 Type=slow timer
  Process::  ID=3 PID=26916 Type=timer
  Process::  ID=4 PID=26917 Type=MI FIFO
  Process::  ID=5 PID=26918 Type=ctl handler
  Process::  ID=6 PID=26919 Type=tcp receiver child=0
  Process::  ID=7 PID=26920 Type=tcp main process


Which process should I attach? udp receiver?

I've tryed to attach udp receiver process in gdb, and by pressing
"next", "next", "next" (running gdb in the kamailio sources directory)
I see nothing about dns functions. Could you please give me some
indications please?

Thanks a lot.





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


From daniel at kamailio.org  Fri Jun 10 23:23:41 2011
From: daniel at kamailio.org (Daniel-Constantin Mierla)
Date: Fri, 10 Jun 2011 23:23:41 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
Message-ID: <4DF28B5D.9010404@kamailio.org>

Hello,

On 6/10/11 11:16 PM, I?aki Baz Castillo wrote:
> 2011/6/10 Daniel-Constantin Mierla<miconda at gmail.com>:
>> Could you set children to 1, attach with gdb to the sip worker handling the
>> invite, send the call and execute step by step to see which condition fails?
>> A simple config with t_relay() should make the debugging easier.
> Hi Daniel. Which worker should I attach? My kamailio listens in UDP
> and TCP. I send the request via UDP and it should do NAPTR stuf and
> choose SIP TCP (best priority). So:
>
>    Process::  ID=0 PID=26913 Type=attendant
>    Process::  ID=1 PID=26914 Type=udp receiver child=0 sock=192.168.1.16:8080
>    Process::  ID=2 PID=26915 Type=slow timer
>    Process::  ID=3 PID=26916 Type=timer
>    Process::  ID=4 PID=26917 Type=MI FIFO
>    Process::  ID=5 PID=26918 Type=ctl handler
>    Process::  ID=6 PID=26919 Type=tcp receiver child=0
>    Process::  ID=7 PID=26920 Type=tcp main process
>
>
> Which process should I attach? udp receiver?
>
> I've tryed to attach udp receiver process in gdb, and by pressing
> "next", "next", "next" (running gdb in the kamailio sources directory)
> I see nothing about dns functions. Could you please give me some
> indications please?
yes, attach to the process that receives the sip request, in this case 
the udp receiver.

Then if you use a simple config with t_relay(), go to tm module, see 
where t_relay() is defined, follow a bit the code and set a breakpoint 
by file and line number at a convenient location (as much as close to 
dns lookup functions if you can spot them in the code), so you don't do 
next/next/ too many times.

Cheers.
Daniel


From ibc at aliax.net  Fri Jun 10 23:26:46 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 10 Jun 2011 23:26:46 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF28B5D.9010404@kamailio.org>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
Message-ID: <BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>

2011/6/10 Daniel-Constantin Mierla <daniel at kamailio.org>:
> yes, attach to the process that receives the sip request, in this case the
> udp receiver.
>
> Then if you use a simple config with t_relay(), go to tm module, see where
> t_relay() is defined, follow a bit the code and set a breakpoint by file and
> line number at a convenient location (as much as close to dns lookup
> functions if you can spot them in the code), so you don't do next/next/ too
> many times.

Understood. Let me some time to do it and I'll come back. Thanks.

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


From ibc at aliax.net  Sat Jun 11 06:54:32 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sat, 11 Jun 2011 06:54:32 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF26073.30001@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr> <4DF26073.30001@gmail.com>
Message-ID: <BANLkTi=mkPk=2VgRkxqadEnoYOwzB4va2g@mail.gmail.com>

2011/6/10 Daniel-Constantin Mierla <miconda at gmail.com>:
> have you spotted in SIP/SIMPLE specs what has to be sent when the presentity
> is offline? My quick google was not that succesful. Maybe Inaki has the
> specs more fresh indexed in memory and can help.

If the user has not published a "offline" presentity, then the server
must not generate one. Sending an empty NOTIFY is the correct
behaviour, which means "unknown state" (most of the subscribers should
interpret it as  "offline").

What is the purpose of publishing a presentity with "offline" status?
Well, as most of the SIMPLE presence stuff, it's just a way to make
things hyper-complex.

There is however an exception: RFC 4827. The client goes on holidays
for long time and decides to upload,, via *XCAP* a *permanent* (non
expirable as a PUBLISH publication) pidf document in which it set the
info it desires (maybe a status "offline" with a message "I'm on
holiday until June 20").


> I based my assumption that the sip/simple presence server should not
> generate itself a presence/pidf document with status closed/offline since
> the pidf has a <tuple id="..."> node that is generated by presentity device.

Right. The presence server must never generate a pidf document in
behalf of the user.

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


From ibc at aliax.net  Sat Jun 11 06:55:58 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sat, 11 Jun 2011 06:55:58 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF277FD.8050908@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr> <4DF26073.30001@gmail.com>
	<4DF2672F.2080006@southeren.com> <4DF277FD.8050908@gmail.com>
Message-ID: <BANLkTikL_05vz+2kAWkYVH40nE99xoA4mA@mail.gmail.com>

2011/6/10 Daniel-Constantin Mierla <miconda at gmail.com>:
>> Specifically, the RFC says:
>>
>> ?"During composition, a presence agent (PA) may encounter a stored
>> <timed-status> element that covers the present time. ?The PA MAY either
>> discard that element or MAY
>> ? convert it to a regular <status> element if it considers that
>> information more credible."
>>
>> So, if a client uploads a presence document that contains a <timed-status>
>> element, that element could be sent by the PA when the client goes offline,
>> rather than the entire document being discarded (which is what I suspect is
>> happening now).
>
> we don't have this rfc implemented in kamailio at this time. Is any UA you
> know doing this kind of thing?

That is anohter hyper-exotic and ugly SIMPLE-Presence feature.

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


From ibc at aliax.net  Sat Jun 11 09:24:18 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sat, 11 Jun 2011 09:24:18 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
	<BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
Message-ID: <BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>

2011/6/10 I?aki Baz Castillo <ibc at aliax.net>:
>> Then if you use a simple config with t_relay(), go to tm module, see where
>> t_relay() is defined, follow a bit the code and set a breakpoint by file and
>> line number at a convenient location (as much as close to dns lookup
>> functions if you can spot them in the code), so you don't do next/next/ too
>> many times.

Hi Daniel, I tryed to extract some useful data but got nothing. I've
set breakpoints by indicating file:line of tm module as well as
resolve.c file. The same using function names as breakpoints, etc. No
result at all, I just get stuf about "UDP, IO, read" and so.

Anyhow I've make some other tests and concluded that the issue is very
simple: Kamailio performs NAPTR query but completely ignores its
result. I've reported the bug in the tracker:

  http://sip-router.org/tracker/index.php?do=details&task_id=135

Cheers.

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


From craig at southeren.com  Fri Jun 10 23:59:31 2011
From: craig at southeren.com (Craig Southeren)
Date: Fri, 10 Jun 2011 14:59:31 -0700
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <4DF27D07.9020901@gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr>
	<4DF26073.30001@gmail.com> <4DF2672F.2080006@southeren.com>
	<4DF277FD.8050908@gmail.com> <4DF27981.1040706@southeren.com>
	<4DF27D07.9020901@gmail.com>
Message-ID: <4DF293C3.7050408@southeren.com>

On 10/06/2011 1:22 PM, Daniel-Constantin Mierla wrote:
>
>
> On 6/10/11 10:07 PM, Craig Southeren wrote:
>> On 10/06/2011 1:01 PM, Daniel-Constantin Mierla wrote:..deleted
>>>>
>>>> So, if a client uploads a presence document that contains a 
>>>> <timed-status> element, that element could be sent by the PA when 
>>>> the client goes offline, rather than the entire document being 
>>>> discarded (which is what I suspect is happening now).
>>>
>>> we don't have this rfc implemented in kamailio at this time. Is any 
>>> UA you know doing this kind of thing?
>>
>> No
> Then it is the time for Eugen to add it in his sip client and send the 
> patch for enhancing kamailio's presence server with rfc4481, as well  ;-)

Actually, as Ekiga uses Opal for all SIP work, this means that I (or 
Robert) would have to do it :)

    Craig


-- 

-----------------------------------------------------------------------
  Craig Southeren          Post Increment ? VoIP Consulting and Software
  craigs at postincrement.com.au                   www.postincrement.com.au

  Phone:  +61 243654666      ICQ: #86852844
  Fax:    +61 243656905      MSN: craig_southeren at hotmail.com
  Mobile: +61 417231046      Jabber: craigs at jabber.org

  "Science is the poetry of reality." Richard Dawkins



From craig at southeren.com  Sun Jun 12 04:00:10 2011
From: craig at southeren.com (Craig Southeren)
Date: Sat, 11 Jun 2011 19:00:10 -0700
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <BANLkTikL_05vz+2kAWkYVH40nE99xoA4mA@mail.gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>	<4DF0AA1B.9060408@gmail.com>	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>	<4DF1EF85.5070509@gmail.com>	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr>	<4DF26073.30001@gmail.com>	<4DF2672F.2080006@southeren.com>	<4DF277FD.8050908@gmail.com>
	<BANLkTikL_05vz+2kAWkYVH40nE99xoA4mA@mail.gmail.com>
Message-ID: <4DF41DAA.3080605@southeren.com>

On 10/06/2011 9:55 PM, I?aki Baz Castillo wrote:
> 2011/6/10 Daniel-Constantin Mierla<miconda at gmail.com>:
>>> Specifically, the RFC says:
>>>
>>>   "During composition, a presence agent (PA) may encounter a stored
>>> <timed-status>  element that covers the present time.  The PA MAY either
>>> discard that element or MAY
>>>    convert it to a regular<status>  element if it considers that
>>> information more credible."
>>>
>>> So, if a client uploads a presence document that contains a<timed-status>
>>> element, that element could be sent by the PA when the client goes offline,
>>> rather than the entire document being discarded (which is what I suspect is
>>> happening now).
>> we don't have this rfc implemented in kamailio at this time. Is any UA you
>> know doing this kind of thing?
> That is anohter hyper-exotic and ugly SIMPLE-Presence feature.
I agree :)

-- 

-----------------------------------------------------------------------
  Craig Southeren          Post Increment ? VoIP Consulting and Software
  craigs at postincrement.com.au                   www.postincrement.com.au

  Phone:  +61 243654666      ICQ: #86852844
  Fax:    +61 243656905      MSN: craig_southeren at hotmail.com
  Mobile: +61 417231046      Jabber: craigs at jabber.org

  "Science is the poetry of reality." Richard Dawkins



From robertj at voxlucida.com.au  Sat Jun 11 10:24:40 2011
From: robertj at voxlucida.com.au (Robert Jongbloed)
Date: Sat, 11 Jun 2011 18:24:40 +1000
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <BANLkTi=mkPk=2VgRkxqadEnoYOwzB4va2g@mail.gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr> <4DF26073.30001@gmail.com>
	<BANLkTi=mkPk=2VgRkxqadEnoYOwzB4va2g@mail.gmail.com>
Message-ID: <027457B5-0200-4B1A-8BD3-64EDCA59378A@voxlucida.com.au>


On 11/06/2011, at 2:54 PM, I?aki Baz Castillo wrote:

>> I based my assumption that the sip/simple presence server should not
>> generate itself a presence/pidf document with status closed/offline since
>> the pidf has a <tuple id="..."> node that is generated by presentity device.
> 
> Right. The presence server must never generate a pidf document in
> behalf of the user.

Fair enough.

There were some hints in the RFC that the clients should detect "stale" presence themselves. Certainly would not be the purview of the PA.

One thing though, other than immediately after a SUBSCRIBE, will Kamailio ever send an empty NOTiFY? If so, what would trigger it?

--------
Robert Jongbloed
Vox Lucida Pty. Ltd.



From ibc at aliax.net  Sun Jun 12 12:11:56 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sun, 12 Jun 2011 12:11:56 +0200
Subject: [SR-Users] Double tuple in notify
In-Reply-To: <4DF27F5B.5020501@pu-pm.univ-fcomte.fr>
References: <4DF27F5B.5020501@pu-pm.univ-fcomte.fr>
Message-ID: <BANLkTimHoUUztsM0wH89o6bN_HJ0RNM=dg@mail.gmail.com>

The id of both tuples are different so the server must keep both. However
the client sholud use if-match header on the second PUBLISH to ensure it
will replace the existing one.

El 10/06/2011 21:32, "Eugen Dedu" <Eugen.Dedu at pu-pm.univ-fcomte.fr>
escribi?:

Hi again,

 Then it is the time for Eugen to add it in his sip client and send
> the patch for enhancing kamailio's presence server with rfc4481, as
> well
>

:o)

I start my application, and make my own status be shown.  I change my status
back and forth a few times between Away and DoNotDisturb (each time a
publish is sent), the status returned (in notify) is good.  When I change my
status to Online/Available (and only to this), kamailio returns a double
tuple.  Is it normal to have a double tuple?  If yes, which one to choose
(the one with latest timestamp)?  Here it is:

NOTIFY sip:eugen.dedu at 82.238.108.175:**5060<http://sip:eugen.dedu at 82.238.108.175:5060>SIP/2.0
CSeq: 7 NOTIFY
Via: SIP/2.0/UDP 86.64.162.35;branch=**z9hG4bK7634.4d863c14.0
User-Agent: Kamailio (1.5.3-notls (i386/linux))
From: sip:eugen.dedu at ekiga.net;tag=**f85b0bd16aaafa8479586ac9f88b31**98-29fc
Call-ID: 8a0d723b-0f8d-e011-84d7-**0024d693d8e8 at snoopy
To: sip:eugen.dedu at ekiga.net;tag=**7c75723b-0f8d-e011-84d7-**0024d693d8e8
Contact: <sip:86.64.162.35:5060>
Subscription-State: active;expires=270
Event: presence
Content-Length: 938
Content-Type: application/pidf+xml
Max-Forwards: 70

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:**pidf"
xmlns:dm="urn:ietf:params:xml:**ns:pidf:data-model" xmlns:
rpid="urn:ietf:params:xml:ns:**pidf:rpid" entity="pres:eugen.dedu at ekiga.**
net <pres%3Aeugen.dedu at ekiga.net>">
 <tuple id="T00000002">
   <status>
     <basic>open</basic>
   </status>
   <contact priority="1">sip:eugen.dedu@**ekiga.net<sip%3Aeugen.dedu at ekiga.net>
</contact>
   <note>online - </note>
   <timestamp>2011-06-04T13:55:**56+02:00+02:00</timestamp>
 </tuple>

 <tuple xmlns="urn:ietf:params:xml:ns:**pidf" id="T00000001">
   <status>
     <basic>open</basic>
   </status>
   <contact priority="1">sip:eugen.dedu@**ekiga.net<sip%3Aeugen.dedu at ekiga.net>
</contact>
   <note>away - jjk</note>
   <timestamp>2011-06-04T13:54:**37+02:00+02:00</timestamp>
 </tuple>
 <dm:person xmlns:dm="urn:ietf:params:xml:**ns:pidf:data-model"
xmlns:rpid="urn:ietf:params:**xml:ns:pidf:rpid" id="pid1">
   <rpid:activities>
     <rpid:away/>
   </rpid:activities>
 </dm:person>
</presence>

-- 
Eugen

______________________________**_________________
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 spinov_evgeniy at intalisan.com  Sun Jun 12 21:56:14 2011
From: spinov_evgeniy at intalisan.com (Spinov Evgeniy)
Date: Sun, 12 Jun 2011 23:56:14 +0400
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <4DF25D89.9090104@gmail.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com>
Message-ID: <13993171.20110612235614@intalisan.com>

Hello.


DCM> Hello,

DCM> to understand the scenario:
DCM> - first branch has destination uri ($du) set

Exactly.

DCM> - it failed and gets to failure route where you call ds_next_domain() 
DCM> and $du s still the one from first branch?

Exactly. $du is being set by ds_select_dst() before t_relay() before
any branch in main loop. In failure route after ds_next_domain()
variable $du remains with old IP ( previous asterisk node ), while $rd
variable is updated.

DCM> What do you mean that "not any of the nodes receive the packet ..."?

Kamailio is standing before asterisks and relays packets to them. In
case, when there is no ds_next_domain() - packet is routed correctly
to the curtain node, defined by ds_select_dst(). While after
ds_next_domain() none of the 2 nodes receives the packet and call
terminates by the caller with "Request timeout". Looks like t_relay()
sends packet somewhere to blackhole.

Thank you.

DCM> Cheers.
DCM> Daniel

DCM> On 6/10/11 6:43 PM, Evgeniy Spinov wrote:
>> Hello.
>>
>> I had a Kamailio version of 3.0.3 and during this time configured a
>> failover with simple routine. In short like this:
>>
>> if (ds_next_domain()) {
>>       xlog(.....);
>>       if (!t_relay()) {
>>               xlog(.....);
>>               return;
>>       }
>>       return;
>> } else {
>>       t_reply("503");
>> }
>>
>> Then I've updated to 3.1.3 and was happy enough until I've discovered
>> that my failover is not working anymore.
>>
>> What is observed:
>> - It changes $rd variable, but not changing $du variable, is it ok?
>> - Not any of the nodes receive packet on t_relay after ds_next_domain().
>> - In route decision section, where t_on_reply() and t_on_failure() are
>> located I have the same t_relay() and it works fine, cause there is no
>> any ds_next_domains().
>>
>> Is it a bug or I'm doing something wrong?
>>
>> 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




--
Best regards,
 Spinov Evgeniy. Intalisan. Development team.

 Phone:   +971  4 326-22-69     Skype:  karator
 Mobile:  +971 55 774-04-87     ICQ:    136613603
 Website: www.intalisan.com



From omar at 321communications.com  Mon Jun 13 07:05:21 2011
From: omar at 321communications.com (Omar)
Date: Mon, 13 Jun 2011 01:05:21 -0400
Subject: [SR-Users] Operations With Variables $var- does it need to load a
	module
Message-ID: <6C5E1099-D53A-4C6D-A36C-F0DC07D20C00@321communications.com>


In Kamailio 1.4.3

I have the following scenario:

$var(a) = 0
$var(b) = 0
$var(a) = $(tU{s.len});
$var(b) =  $var(a) - 5;

xlog(" $var(b)");

the result $var(b)  is always the same as $var(a)

should i check if some specific module is loaded?
the debugs does not show any error, and $var(a) and $var(b) are unique.

Variables are working, but not the operations with the variables.

regards

O



From michofr at gmail.com  Mon Jun 13 09:12:47 2011
From: michofr at gmail.com (michel freiha)
Date: Mon, 13 Jun 2011 09:12:47 +0200
Subject: [SR-Users] Proxy Address
Message-ID: <BANLkTi=yeD1S6HW3jurBpCR7re7o6rAJqQ@mail.gmail.com>

Dear All,

I'm using kamailio as Proxy server for registration with RTP proxy as relay
server for voice packets...I have an issue that when sending an Invite, the
audio address in SDP (o header)is not RTP Proxy address while if I send
another Invite to kamailio (second Invite), the RTP Proxy address will be
present in SDP sent back by Kamailio...

Anyone has a clue about what could be that matter?

Regards
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From robertj at voxlucida.com.au  Sun Jun 12 12:42:13 2011
From: robertj at voxlucida.com.au (Robert Jongbloed)
Date: Sun, 12 Jun 2011 20:42:13 +1000
Subject: [SR-Users] Double tuple in notify
In-Reply-To: <BANLkTimHoUUztsM0wH89o6bN_HJ0RNM=dg@mail.gmail.com>
References: <4DF27F5B.5020501@pu-pm.univ-fcomte.fr>
	<BANLkTimHoUUztsM0wH89o6bN_HJ0RNM=dg@mail.gmail.com>
Message-ID: <FE4A99A5-F855-4989-ADA6-F512EC477D83@voxlucida.com.au>

Yes, correct. I don't think this is anything to do with Kamailio, I will detach you from the thread,

Thanks for all your assistance!

--------
Robert Jongbloed
Vox Lucida Pty. Ltd.


On 12/06/2011, at 8:11 PM, I?aki Baz Castillo wrote:

> The id of both tuples are different so the server must keep both. However the client sholud use if-match header on the second PUBLISH to ensure it will replace the existing one.
> 
> 
>> El 10/06/2011 21:32, "Eugen Dedu" <Eugen.Dedu at pu-pm.univ-fcomte.fr> escribi?:
>> 
>> Hi again,
>> 
>> Then it is the time for Eugen to add it in his sip client and send
>> the patch for enhancing kamailio's presence server with rfc4481, as
>> well
>> 
>> :o)
>> 
>> I start my application, and make my own status be shown.  I change my status back and forth a few times between Away and DoNotDisturb (each time a publish is sent), the status returned (in notify) is good.  When I change my status to Online/Available (and only to this), kamailio returns a double tuple.  Is it normal to have a double tuple?  If yes, which one to choose (the one with latest timestamp)?  Here it is:
>> 
>> NOTIFY sip:eugen.dedu at 82.238.108.175:5060 SIP/2.0
>> CSeq: 7 NOTIFY
>> Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK7634.4d863c14.0
>> User-Agent: Kamailio (1.5.3-notls (i386/linux))
>> From: sip:eugen.dedu at ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-29fc
>> Call-ID: 8a0d723b-0f8d-e011-84d7-0024d693d8e8 at snoopy
>> To: sip:eugen.dedu at ekiga.net;tag=7c75723b-0f8d-e011-84d7-0024d693d8e8
>> Contact: <sip:86.64.162.35:5060>
>> Subscription-State: active;expires=270
>> Event: presence
>> Content-Length: 938
>> Content-Type: application/pidf+xml
>> Max-Forwards: 70
>> 
>> <?xml version="1.0" encoding="UTF-8"?>
>> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:
>> rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:eugen.dedu at ekiga.net">
>>  <tuple id="T00000002">
>>    <status>
>>      <basic>open</basic>
>>    </status>
>>    <contact priority="1">sip:eugen.dedu at ekiga.net</contact>
>>    <note>online - </note>
>>    <timestamp>2011-06-04T13:55:56+02:00+02:00</timestamp>
>>  </tuple>
>> 
>>  <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="T00000001">
>>    <status>
>>      <basic>open</basic>
>>    </status>
>>    <contact priority="1">sip:eugen.dedu at ekiga.net</contact>
>>    <note>away - jjk</note>
>>    <timestamp>2011-06-04T13:54:37+02:00+02:00</timestamp>
>>  </tuple>
>>  <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="pid1">
>>    <rpid:activities>
>>      <rpid:away/>
>>    </rpid:activities>
>>  </dm:person>
>> </presence>
>> 
>> -- 
>> Eugen
>> 
>> _______________________________________________
>> 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  Mon Jun 13 10:49:22 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Mon, 13 Jun 2011 10:49:22 +0200
Subject: [SR-Users] Proxy Address
In-Reply-To: <BANLkTi=yeD1S6HW3jurBpCR7re7o6rAJqQ@mail.gmail.com>
References: <BANLkTi=yeD1S6HW3jurBpCR7re7o6rAJqQ@mail.gmail.com>
Message-ID: <4DF5CF12.8020704@pernau.at>

What is the IP address if the first case? A wrong one or the original IP 
address provided by the caller?

If it is the original IP address, either force_rtp_proxy was not called 
or there was a problem wiht the rtpproxy.

Use xlog statements in front and after the rtpproxy functions and verify 
that rtpproxy is called.

regards
klaus

On 13.06.2011 09:12, michel freiha wrote:
> Dear All,
>
> I'm using kamailio as Proxy server for registration with RTP proxy as
> relay server for voice packets...I have an issue that when sending an
> Invite, the audio address in SDP (o header)is not RTP Proxy address
> while if I send another Invite to kamailio (second Invite), the RTP
> Proxy address will be present in SDP sent back by Kamailio...
>
> Anyone has a clue about what could be that matter?
>
> Regards
>
>
>
> _______________________________________________
> 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  Mon Jun 13 10:49:29 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 13 Jun 2011 10:49:29 +0200
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <13993171.20110612235614@intalisan.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com>
	<13993171.20110612235614@intalisan.com>
Message-ID: <4DF5CF19.7090205@gmail.com>

Hello,

On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
> Hello.
>
>
> DCM>  Hello,
>
> DCM>  to understand the scenario:
> DCM>  - first branch has destination uri ($du) set
>
> Exactly.
>
> DCM>  - it failed and gets to failure route where you call ds_next_domain()
> DCM>  and $du s still the one from first branch?
>
> Exactly. $du is being set by ds_select_dst() before t_relay() before
> any branch in main loop. In failure route after ds_next_domain()
> variable $du remains with old IP ( previous asterisk node ), while $rd
> variable is updated.
is any particular reason to use ds_next_domain() after ds_select_dst()? 
ds_next_domain() is the pair for ds_select_domain, in older versions it 
happened that $du was reset automatically internally even by 
ds_next_domain() due to execution of a core function which was no longer 
needed in 3.1.

A quick solution if you need ds_select_dst() in combination with 
ds_next_domain() is to call resetdsturi() in failure route. I will think 
about and see if it is consistent to call it internally.

>
> DCM>  What do you mean that "not any of the nodes receive the packet ..."?
>
> Kamailio is standing before asterisks and relays packets to them. In
> case, when there is no ds_next_domain() - packet is routed correctly
> to the curtain node, defined by ds_select_dst(). While after
> ds_next_domain() none of the 2 nodes receives the packet and call
> terminates by the caller with "Request timeout". Looks like t_relay()
> sends packet somewhere to blackhole.

Can you look at sip trace with ngrep to see where they are sent? 
Probably to the first destination that failed.

Cheers,
Daniel

>
> Thank you.
>
> DCM>  Cheers.
> DCM>  Daniel
>
> DCM>  On 6/10/11 6:43 PM, Evgeniy Spinov wrote:
>>> Hello.
>>>
>>> I had a Kamailio version of 3.0.3 and during this time configured a
>>> failover with simple routine. In short like this:
>>>
>>> if (ds_next_domain()) {
>>>        xlog(.....);
>>>        if (!t_relay()) {
>>>                xlog(.....);
>>>                return;
>>>        }
>>>        return;
>>> } else {
>>>        t_reply("503");
>>> }
>>>
>>> Then I've updated to 3.1.3 and was happy enough until I've discovered
>>> that my failover is not working anymore.
>>>
>>> What is observed:
>>> - It changes $rd variable, but not changing $du variable, is it ok?
>>> - Not any of the nodes receive packet on t_relay after ds_next_domain().
>>> - In route decision section, where t_on_reply() and t_on_failure() are
>>> located I have the same t_relay() and it works fine, cause there is no
>>> any ds_next_domains().
>>>
>>> Is it a bug or I'm doing something wrong?
>>>
>>> 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
>
>
>
> --
> Best regards,
>   Spinov Evgeniy. Intalisan. Development team.
>
>   Phone:   +971  4 326-22-69     Skype:  karator
>   Mobile:  +971 55 774-04-87     ICQ:    136613603
>   Website: www.intalisan.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

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From michofr at gmail.com  Mon Jun 13 11:40:26 2011
From: michofr at gmail.com (michel freiha)
Date: Mon, 13 Jun 2011 11:40:26 +0200
Subject: [SR-Users] Proxy Address
In-Reply-To: <4DF5CF12.8020704@pernau.at>
References: <BANLkTi=yeD1S6HW3jurBpCR7re7o6rAJqQ@mail.gmail.com>
	<4DF5CF12.8020704@pernau.at>
Message-ID: <BANLkTi=Qo1qU4dQoL3GUnvQu+0v5ogVaNw@mail.gmail.com>

Dear Sir,

Can you please have a look to my config file and let me know please why the
rtp proxy is not forced in SDP?


PasteBin address:

http://pastebin.com/NrJR8geQ

Regards

On Mon, Jun 13, 2011 at 10:49 AM, Klaus Darilion <
klaus.mailinglists at pernau.at> wrote:

> What is the IP address if the first case? A wrong one or the original IP
> address provided by the caller?
>
> If it is the original IP address, either force_rtp_proxy was not called or
> there was a problem wiht the rtpproxy.
>
> Use xlog statements in front and after the rtpproxy functions and verify
> that rtpproxy is called.
>
> regards
> klaus
>
>
> On 13.06.2011 09:12, michel freiha wrote:
>
>> Dear All,
>>
>> I'm using kamailio as Proxy server for registration with RTP proxy as
>> relay server for voice packets...I have an issue that when sending an
>> Invite, the audio address in SDP (o header)is not RTP Proxy address
>> while if I send another Invite to kamailio (second Invite), the RTP
>> Proxy address will be present in SDP sent back by Kamailio...
>>
>> Anyone has a clue about what could be that matter?
>>
>> Regards
>>
>>
>>
>> _______________________________________________
>> 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 spinov_evgeniy at intalisan.com  Mon Jun 13 12:03:24 2011
From: spinov_evgeniy at intalisan.com (Evgeniy Spinov)
Date: Mon, 13 Jun 2011 11:03:24 +0100
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <4DF5CF19.7090205@gmail.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com> <13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
Message-ID: <1307959404.2074.26.camel@localhost.localdomain>

On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
> Hello,
> 
> On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
> > Hello.
> >
> >
> > DCM>  Hello,
> >
> > DCM>  to understand the scenario:
> > DCM>  - first branch has destination uri ($du) set
> >
> > Exactly.
> >
> > DCM>  - it failed and gets to failure route where you call ds_next_domain()
> > DCM>  and $du s still the one from first branch?
> >
> > Exactly. $du is being set by ds_select_dst() before t_relay() before
> > any branch in main loop. In failure route after ds_next_domain()
> > variable $du remains with old IP ( previous asterisk node ), while $rd
> > variable is updated.
> is any particular reason to use ds_next_domain() after ds_select_dst()? 
> ds_next_domain() is the pair for ds_select_domain, in older versions it 
> happened that $du was reset automatically internally even by 
> ds_next_domain() due to execution of a core function which was no longer 
> needed in 3.1.

I do not mind using ds_select_dst() however it seems to me, that in
following scenario it will give me a false result:
1. Call 1 is coming 
2. Routed to GW 1
3. Call 2 is coming 
4. Routed to GW 2
5. Call 1 timed out to GW 1 and call is going to failure route, where
ds_select_dst() is being called.

I guess ds_select_dst in this case will give GW 1 again in case of
round-robin fashion.

Also, there is no end of gateways, as ds_select_dst() will always return
one of the gateways, while when no gateways are alive - appropriate
reaction should happen.

That's why ds_next_domain() is more suitable, as performing all the
required actions without magic. Performed at least.


> A quick solution if you need ds_select_dst() in combination with 
> ds_next_domain() is to call resetdsturi() in failure route. I will think 
> about and see if it is consistent to call it internally.

Works for me.

> 
> >
> > DCM>  What do you mean that "not any of the nodes receive the packet ..."?
> >
> > Kamailio is standing before asterisks and relays packets to them. In
> > case, when there is no ds_next_domain() - packet is routed correctly
> > to the curtain node, defined by ds_select_dst(). While after
> > ds_next_domain() none of the 2 nodes receives the packet and call
> > terminates by the caller with "Request timeout". Looks like t_relay()
> > sends packet somewhere to blackhole.
> 
> Can you look at sip trace with ngrep to see where they are sent? 
> Probably to the first destination that failed.

Yes. It sends packet to previous gateway, while in INVITE another IP
mentioned ( look at 1 and 2 strings ):

13:59:36.533705 IP KAMAILIO_IP.sip > GW1_IP.sip: UDP, length 1020
E....H.. at .4..(...(.........UINVITE sip:*44 at GW2_IP:5060 SIP/2.0
Record-Route:
<sip:KAMAILIO_IP;lr=on;ftag=ustdc;did=c4e.a81165f4;nat=yes>
Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bKccae.c54a1682.1
Max-Forwards: 69
To: <sip:*44 at KAMAILIO_IP>
From: "2_101" <sip:2_101 at KAMAILIO_IP>;tag=ustdc
Call-ID: yuwofwddoaqozwh at localhost.localdomain
CSeq: 527 INVITE
Contact: <sip:2_101 at UAC_PUBLIC_IP:5061>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 330



Looks like a bug.

> 
> Cheers,
> Daniel
> 
> >
> > Thank you.
> >
> > DCM>  Cheers.
> > DCM>  Daniel
> >
> > DCM>  On 6/10/11 6:43 PM, Evgeniy Spinov wrote:
> >>> Hello.
> >>>
> >>> I had a Kamailio version of 3.0.3 and during this time configured a
> >>> failover with simple routine. In short like this:
> >>>
> >>> if (ds_next_domain()) {
> >>>        xlog(.....);
> >>>        if (!t_relay()) {
> >>>                xlog(.....);
> >>>                return;
> >>>        }
> >>>        return;
> >>> } else {
> >>>        t_reply("503");
> >>> }
> >>>
> >>> Then I've updated to 3.1.3 and was happy enough until I've discovered
> >>> that my failover is not working anymore.
> >>>
> >>> What is observed:
> >>> - It changes $rd variable, but not changing $du variable, is it ok?
> >>> - Not any of the nodes receive packet on t_relay after ds_next_domain().
> >>> - In route decision section, where t_on_reply() and t_on_failure() are
> >>> located I have the same t_relay() and it works fine, cause there is no
> >>> any ds_next_domains().
> >>>
> >>> Is it a bug or I'm doing something wrong?
> >>>
> >>> 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
> >
> >
> >
> > _______________________________________________
> > 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  Mon Jun 13 12:04:17 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 13 Jun 2011 12:04:17 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
	<BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
	<BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>
Message-ID: <4DF5E0A1.1080605@gmail.com>

Hello,

On 6/11/11 9:24 AM, I?aki Baz Castillo wrote:
> 2011/6/10 I?aki Baz Castillo<ibc at aliax.net>:
>>> Then if you use a simple config with t_relay(), go to tm module, see where
>>> t_relay() is defined, follow a bit the code and set a breakpoint by file and
>>> line number at a convenient location (as much as close to dns lookup
>>> functions if you can spot them in the code), so you don't do next/next/ too
>>> many times.
> Hi Daniel, I tryed to extract some useful data but got nothing. I've
> set breakpoints by indicating file:line of tm module as well as
> resolve.c file. The same using function names as breakpoints, etc. No
> result at all, I just get stuf about "UDP, IO, read" and so.
>
> Anyhow I've make some other tests and concluded that the issue is very
> simple: Kamailio performs NAPTR query but completely ignores its
> result. I've reported the bug in the tracker:
>
>    http://sip-router.org/tracker/index.php?do=details&task_id=135
I saw Andrei jumped in and added case insensitive comparison of naptr 
flags -- just to conclude this discussion, is it working on now?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Mon Jun 13 12:12:38 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 13 Jun 2011 12:12:38 +0200
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <1307959404.2074.26.camel@localhost.localdomain>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com>
	<13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
Message-ID: <4DF5E296.6070607@gmail.com>



On 6/13/11 12:03 PM, Evgeniy Spinov wrote:
> On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
>>> Hello.
>>>
>>>
>>> DCM>   Hello,
>>>
>>> DCM>   to understand the scenario:
>>> DCM>   - first branch has destination uri ($du) set
>>>
>>> Exactly.
>>>
>>> DCM>   - it failed and gets to failure route where you call ds_next_domain()
>>> DCM>   and $du s still the one from first branch?
>>>
>>> Exactly. $du is being set by ds_select_dst() before t_relay() before
>>> any branch in main loop. In failure route after ds_next_domain()
>>> variable $du remains with old IP ( previous asterisk node ), while $rd
>>> variable is updated.
>> is any particular reason to use ds_next_domain() after ds_select_dst()?
>> ds_next_domain() is the pair for ds_select_domain, in older versions it
>> happened that $du was reset automatically internally even by
>> ds_next_domain() due to execution of a core function which was no longer
>> needed in 3.1.
> I do not mind using ds_select_dst() however it seems to me, that in
> following scenario it will give me a false result:
> 1. Call 1 is coming
> 2. Routed to GW 1
> 3. Call 2 is coming
> 4. Routed to GW 2
> 5. Call 1 timed out to GW 1 and call is going to failure route, where
> ds_select_dst() is being called.
> I guess ds_select_dst in this case will give GW 1 again in case of
> round-robin fashion.
>
> Also, there is no end of gateways, as ds_select_dst() will always return
> one of the gateways, while when no gateways are alive - appropriate
> reaction should happen.
>
> That's why ds_next_domain() is more suitable, as performing all the
> required actions without magic. Performed at least.
ds_select_dst()/ds_next_dst() and ds_select_domain()/ds_next_domain() 
consume the already-tried destinations -- in any of the cases, none of 
gateways will be tried two times.


ds_*_dst() sets $du and $ds_*_domain() sets $ru -- this is the 
difference between them.

When you do first ds_select_dst() you set $du and when then you do 
$ds_next_domain() you set $ru -- if you would do $ds_next_dst() then $du 
will be set with the new gateway address.

>> A quick solution if you need ds_select_dst() in combination with
>> ds_next_domain() is to call resetdsturi() in failure route. I will think
>> about and see if it is consistent to call it internally.
> Works for me.
>
>>> DCM>   What do you mean that "not any of the nodes receive the packet ..."?
>>>
>>> Kamailio is standing before asterisks and relays packets to them. In
>>> case, when there is no ds_next_domain() - packet is routed correctly
>>> to the curtain node, defined by ds_select_dst(). While after
>>> ds_next_domain() none of the 2 nodes receives the packet and call
>>> terminates by the caller with "Request timeout". Looks like t_relay()
>>> sends packet somewhere to blackhole.
>> Can you look at sip trace with ngrep to see where they are sent?
>> Probably to the first destination that failed.
> Yes. It sends packet to previous gateway, while in INVITE another IP
> mentioned ( look at 1 and 2 strings ):
>
> 13:59:36.533705 IP KAMAILIO_IP.sip>  GW1_IP.sip: UDP, length 1020
> E....H.. at .4..(...(.........UINVITE sip:*44 at GW2_IP:5060 SIP/2.0
> Record-Route:
> <sip:KAMAILIO_IP;lr=on;ftag=ustdc;did=c4e.a81165f4;nat=yes>
> Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bKccae.c54a1682.1
> Max-Forwards: 69
> To:<sip:*44 at KAMAILIO_IP>
> From: "2_101"<sip:2_101 at KAMAILIO_IP>;tag=ustdc
> Call-ID: yuwofwddoaqozwh at localhost.localdomain
> CSeq: 527 INVITE
> Contact:<sip:2_101 at UAC_PUBLIC_IP:5061>
> Content-Type: application/sdp
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Supported: replaces,norefersub,100rel
> User-Agent: Twinkle/1.4.2
> Content-Length: 330
>
>
>
> Looks like a bug.

This is not a bug in sip routing as long as $du is set, then it is the 
field used for sip routing no matter you have in $ru -- $du is the 
internal outbound proxy address overwriting request-uri address.

Maybe will be safe to introduce resetdsturi() in case of 
ds_next_domain() to be safe in combinations like 
ds_select_dst()+ds_next_domain().

Cheers,
Daniel



-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Mon Jun 13 12:16:59 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 13 Jun 2011 12:16:59 +0200
Subject: [SR-Users] Meaning of empty body in NOTIFY
In-Reply-To: <BANLkTi=mkPk=2VgRkxqadEnoYOwzB4va2g@mail.gmail.com>
References: <4DEBE787.2080407@pu-pm.univ-fcomte.fr>
	<4DEFD63F.5080109@pu-pm.univ-fcomte.fr>
	<4DF0AA1B.9060408@gmail.com>
	<4DF1EC3A.3060507@pu-pm.univ-fcomte.fr>
	<4DF1EF85.5070509@gmail.com>
	<4DF1F1AA.6010800@pu-pm.univ-fcomte.fr>
	<3AA68EEA-936F-4FEB-8588-9710A3DBE462@gmail.com>
	<4DF25B2C.4060906@pu-pm.univ-fcomte.fr> <4DF26073.30001@gmail.com>
	<BANLkTi=mkPk=2VgRkxqadEnoYOwzB4va2g@mail.gmail.com>
Message-ID: <4DF5E39B.9000906@gmail.com>

Hello,

On 6/11/11 6:54 AM, I?aki Baz Castillo wrote:
> 2011/6/10 Daniel-Constantin Mierla<miconda at gmail.com>:
>> have you spotted in SIP/SIMPLE specs what has to be sent when the presentity
>> is offline? My quick google was not that succesful. Maybe Inaki has the
>> specs more fresh indexed in memory and can help.
> If the user has not published a "offline" presentity, then the server
> must not generate one. Sending an empty NOTIFY is the correct
> behaviour, which means "unknown state" (most of the subscribers should
> interpret it as  "offline").
>
> What is the purpose of publishing a presentity with "offline" status?
> Well, as most of the SIMPLE presence stuff, it's just a way to make
> things hyper-complex.
>
> There is however an exception: RFC 4827. The client goes on holidays
> for long time and decides to upload,, via *XCAP* a *permanent* (non
> expirable as a PUBLISH publication) pidf document in which it set the
> info it desires (maybe a status "offline" with a message "I'm on
> holiday until June 20").
right, it was the rfc i wanted to search for -- this features in 
kamailio presence implementation, can be controlled via module parameter 
pidf_manipulation , see:
http://kamailio.org/docs/modules/stable/modules_k/presence_xml.html#id2940723

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From spinov_evgeniy at intalisan.com  Mon Jun 13 12:28:49 2011
From: spinov_evgeniy at intalisan.com (Evgeniy Spinov)
Date: Mon, 13 Jun 2011 11:28:49 +0100
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <4DF5E296.6070607@gmail.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com> <13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
	<4DF5E296.6070607@gmail.com>
Message-ID: <1307960929.6124.3.camel@localhost.localdomain>

On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:
> 
> On 6/13/11 12:03 PM, Evgeniy Spinov wrote:
> > On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
> >> Hello,
> >>
> >> On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
> >>> Hello.
> >>>
> >>>
> >>> DCM>   Hello,
> >>>
> >>> DCM>   to understand the scenario:
> >>> DCM>   - first branch has destination uri ($du) set
> >>>
> >>> Exactly.
> >>>
> >>> DCM>   - it failed and gets to failure route where you call ds_next_domain()
> >>> DCM>   and $du s still the one from first branch?
> >>>
> >>> Exactly. $du is being set by ds_select_dst() before t_relay() before
> >>> any branch in main loop. In failure route after ds_next_domain()
> >>> variable $du remains with old IP ( previous asterisk node ), while $rd
> >>> variable is updated.
> >> is any particular reason to use ds_next_domain() after ds_select_dst()?
> >> ds_next_domain() is the pair for ds_select_domain, in older versions it
> >> happened that $du was reset automatically internally even by
> >> ds_next_domain() due to execution of a core function which was no longer
> >> needed in 3.1.
> > I do not mind using ds_select_dst() however it seems to me, that in
> > following scenario it will give me a false result:
> > 1. Call 1 is coming
> > 2. Routed to GW 1
> > 3. Call 2 is coming
> > 4. Routed to GW 2
> > 5. Call 1 timed out to GW 1 and call is going to failure route, where
> > ds_select_dst() is being called.
> > I guess ds_select_dst in this case will give GW 1 again in case of
> > round-robin fashion.
> >
> > Also, there is no end of gateways, as ds_select_dst() will always return
> > one of the gateways, while when no gateways are alive - appropriate
> > reaction should happen.
> >
> > That's why ds_next_domain() is more suitable, as performing all the
> > required actions without magic. Performed at least.
> ds_select_dst()/ds_next_dst() and ds_select_domain()/ds_next_domain() 
> consume the already-tried destinations -- in any of the cases, none of 
> gateways will be tried two times.
> 
> 
> ds_*_dst() sets $du and $ds_*_domain() sets $ru -- this is the 
> difference between them.
> 
> When you do first ds_select_dst() you set $du and when then you do 
> $ds_next_domain() you set $ru -- if 

> you would do $ds_next_dst() then $du 
> will be set with the new gateway address.

Changed to ds_next_dst() and it worked. The one thing, which confused me
is that this was working before 3.1 in 3.0 version. 

Thanks for detailed explanation. Now I've got it.


> 
> >> A quick solution if you need ds_select_dst() in combination with
> >> ds_next_domain() is to call resetdsturi() in failure route. I will think
> >> about and see if it is consistent to call it internally.
> > Works for me.
> >
> >>> DCM>   What do you mean that "not any of the nodes receive the packet ..."?
> >>>
> >>> Kamailio is standing before asterisks and relays packets to them. In
> >>> case, when there is no ds_next_domain() - packet is routed correctly
> >>> to the curtain node, defined by ds_select_dst(). While after
> >>> ds_next_domain() none of the 2 nodes receives the packet and call
> >>> terminates by the caller with "Request timeout". Looks like t_relay()
> >>> sends packet somewhere to blackhole.
> >> Can you look at sip trace with ngrep to see where they are sent?
> >> Probably to the first destination that failed.
> > Yes. It sends packet to previous gateway, while in INVITE another IP
> > mentioned ( look at 1 and 2 strings ):
> >
> > 13:59:36.533705 IP KAMAILIO_IP.sip>  GW1_IP.sip: UDP, length 1020
> > E....H.. at .4..(...(.........UINVITE sip:*44 at GW2_IP:5060 SIP/2.0
> > Record-Route:
> > <sip:KAMAILIO_IP;lr=on;ftag=ustdc;did=c4e.a81165f4;nat=yes>
> > Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bKccae.c54a1682.1
> > Max-Forwards: 69
> > To:<sip:*44 at KAMAILIO_IP>
> > From: "2_101"<sip:2_101 at KAMAILIO_IP>;tag=ustdc
> > Call-ID: yuwofwddoaqozwh at localhost.localdomain
> > CSeq: 527 INVITE
> > Contact:<sip:2_101 at UAC_PUBLIC_IP:5061>
> > Content-Type: application/sdp
> > Allow:
> > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> > Supported: replaces,norefersub,100rel
> > User-Agent: Twinkle/1.4.2
> > Content-Length: 330
> >
> >
> >
> > Looks like a bug.
> 
> This is not a bug in sip routing as long as $du is set, then it is the 
> field used for sip routing no matter you have in $ru -- $du is the 
> internal outbound proxy address overwriting request-uri address.
> 
> Maybe will be safe to introduce resetdsturi() in case of 
> ds_next_domain() to be safe in combinations like 
> ds_select_dst()+ds_next_domain().

I guess mention of this will be welcome in docs. Cause I've got my
working example somewhere from Internet. I guess kamailio beginners may
do the same and it will not work without any visible reason.


> 
> Cheers,
> Daniel
> 
> 
> 




From ibc at aliax.net  Mon Jun 13 14:48:18 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 13 Jun 2011 14:48:18 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF5E0A1.1080605@gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
	<BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
	<BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>
	<4DF5E0A1.1080605@gmail.com>
Message-ID: <BANLkTinqSkPH=JtACxqaoEQU05eZk35oCg@mail.gmail.com>

2011/6/13 Daniel-Constantin Mierla <miconda at gmail.com>:
> I saw Andrei jumped in and added case insensitive comparison of naptr flags
> -- just to conclude this discussion, is it working on now?

Hi Daniel, I've tested it right *now* :)
Yes, it works.

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


From michofr at gmail.com  Mon Jun 13 15:37:22 2011
From: michofr at gmail.com (michel freiha)
Date: Mon, 13 Jun 2011 15:37:22 +0200
Subject: [SR-Users] RTP proxy
Message-ID: <BANLkTimtyz88ehhj6finEcn42ZobkxtB4w@mail.gmail.com>

Dear All,

I'm using the below config plan for routing my calls...The issue that
forcing rtp proxy is not working well and the rtp proxy is never
forced....Can you please provide me a guidance here?


        if($rU=~"^00.*" )
        {


                if(!cr_route("default", "domain.com", "$rU", "$rU",
"call_id")){
                  sl_send_reply("403", "Not allowed");
                } else {

                setbflag(6);
                route(1);
                }
                exit; ## this was missing here -bogdan!!!
        }
        else
        {
                route(LOCATION);

                route(RELAY);
        }
}



############# THIS IS THE GENERIC RELAY ROUTE THAT MUST BE USED ALL THE TIME
FOR RELAY
route[1]
{


        # apply any NAT changes?
        if(isflagset(5)||isbflagset(6)) {
                route(4);
        }



        exit;
}

route[4]
{


        if (is_method("BYE|CANCEL")) {
                unforce_rtp_proxy();
        } else if (is_method("INVITE")){

                if (nat_uac_test("8")) {
                    *    force_rtp_proxy();*
                } else {
                        force_rtp_proxy();
                }
                xlog("L_ERR","66666666666666666666666666666666666666666");
                t_on_failure("1");
                t_on_reply("1");



        };

The call is going through Route[1] then through Route[4], but rtp proxy is
never forced...Any tips please?

Regards
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From miconda at gmail.com  Mon Jun 13 15:55:15 2011
From: miconda at gmail.com (Daniel-Constantine Mierla)
Date: Mon, 13 Jun 2011 15:55:15 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTinqSkPH=JtACxqaoEQU05eZk35oCg@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
	<BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
	<BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>
	<4DF5E0A1.1080605@gmail.com>
	<BANLkTinqSkPH=JtACxqaoEQU05eZk35oCg@mail.gmail.com>
Message-ID: <0847B1C4-F998-4471-9870-52E247928CBD@gmail.com>



On Jun 13, 2011, at 2:48 PM, I?aki Baz Castillo <ibc at aliax.net> wrote:

> 2011/6/13 Daniel-Constantin Mierla <miconda at gmail.com>:
>> I saw Andrei jumped in and added case insensitive comparison of naptr flags
>> -- just to conclude this discussion, is it working on now?
> 
> Hi Daniel, I've tested it right *now* :)
> Yes, it works.

Ok, thanks.

Cheers,
Daniel


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


From mino.haluz at gmail.com  Mon Jun 13 16:29:29 2011
From: mino.haluz at gmail.com (Mino Haluz)
Date: Mon, 13 Jun 2011 16:29:29 +0200
Subject: [SR-Users] Multiple INVITEs and discarded call_control
In-Reply-To: <4DED35BA.8020901@gmail.com>
References: <BANLkTim_pCyDB8ecTkzRbT6EjniTZpit7g@mail.gmail.com>
	<4DED35BA.8020901@gmail.com>
Message-ID: <BANLkTikwqAJYJLTFmiiKFM1YA9WYLHk-rw@mail.gmail.com>

It does not work, t_newtran always returns success, so it will never
absorb the retransmission.
So what I did was:

           if ($sht(a=>$ci::retrans) == 0) {
               $sht(a=>$ci::retrans) = 1;
           } else {
               exit();
           }

          xlog("L_INFO","XLOG: xxx");
          call_control();

I prepared sipp scenario which generates 3 INVITEs separated by 200ms.
It will gives me this:

Jun 13 16:07:04 no-testing /sbin/kamailio[5274]: INFO: <script>: XLOG: xxx
Jun 13 16:07:04 no-testing /sbin/kamailio[5273]: WARNING: call_control
[call_control.c:1156]: dialog to trace controlled call was not
created. discarding callcontrol.
Jun 13 16:07:04 no-testing /sbin/kamailio[5275]: WARNING: call_control
[call_control.c:1156]: dialog to trace controlled call was not
created. discarding callcontrol.

So, the first INVITE was processed by call_control, another 2 were
not. But WHAT executed those two call_control is some kind of miracle
for me ... Callcontrol module is somehow executed at the beginning of
the config file where the values are set or why?

This is what I always set at the beginning:

modparam("dialog", "dlg_flag", 4)
modparam("call_control", "diverter_avp_id", 256)                #used
to set billing party. default is From uri
modparam("call_control", "canonical_uri_avp", "$avp(s:can_uri)")



On Mon, Jun 6, 2011 at 10:16 PM, Daniel-Constantin Mierla
<miconda at gmail.com> wrote:
> Hello,
>
> if it is a retransmission, then just do t_newtran() in your config file
> before handling it to call control to be sure the retransmitted requests are
> absorbed.
> http://kamailio.org/docs/modules/stable/modules/tm.html#t_newtran
>
> Cheers,
> Daniel
>
> On 6/6/11 4:29 PM, Mino Haluz wrote:
>
> Hi,
> my kamailio server is receiving from some customers 3 identical INVITEs when
> call is initiated (separated by 200ms). Those 3 INVITEs are making a big
> problem with call_control:
> WARNING: call_control [call_control.c:1156]: dialog to trace controlled call
> was not created. discarding callcontrol.
> That is why, the prepaid limit is not working at all in this case. This way
> the user can hack the prepaid protection of the account. Otherwise the
> call_control is fuilly functional.
> Anybody experienced the similar problem? If so, how to resolve it?
> Thanks,
> Mino Haluz
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda


From doctore96 at gmail.com  Mon Jun 13 17:09:34 2011
From: doctore96 at gmail.com (jose luis millan)
Date: Mon, 13 Jun 2011 17:09:34 +0200
Subject: [SR-Users] Use of mtree module having multiple trees in the same
	table
Message-ID: <BANLkTik+dR8iBdrr1vtWfjzdZXpsujWzWw@mail.gmail.com>

hello,

I wonder if anyone could help me understanding how can I use this
module in a scenario where multiple mtrees are defined behind the same
database table.

I can't guess how to speficy to 'mt_match' the mtree I want to match
against, since the 'mtree' module parameter defines a specific memory
tree based on a specific table, but not based on a specific 'tname'
value.


Any help is apreciated,

Thanks in advance.


From realjijo at gmail.com  Tue Jun 14 01:54:07 2011
From: realjijo at gmail.com (Jijo)
Date: Mon, 13 Jun 2011 19:54:07 -0400
Subject: [SR-Users] Fwd: 200OK Retransmission Error
In-Reply-To: <BANLkTi=6ngJ=FXkGnBj6P0C9001UgL6ymQ@mail.gmail.com>
References: <BANLkTi=6ngJ=FXkGnBj6P0C9001UgL6ymQ@mail.gmail.com>
Message-ID: <BANLkTi=Sj0x0AMNzWJGLf-76aZECL-wQZQ@mail.gmail.com>

Hi All,

In an INVITE transaction, no ACK from A party and B party continue
retransmission of 200 OK. I have observed that kamailio is not passing the
retransmitted 200 OK message from B party to the Application(script) after 3
seconds. Instead kamailio directly sending to A party without passing to the
script.

I see in the logs that kamailio is cleaning up the transaction after 3
seconds..look for the string "DEBUG: t_lookup_request: no transaction
found"  before this log it is cleaning up the timer in the log "DEBUG:
cleanup_uac_timers: RETR/FR timers reset"

What does it mean by that??
How can we make the kamailio to pass the response to the application?

Please let me know your comments.. Here is the flow. I cannot send the
complete logs which is quite big. so i grep'd using the stirng
"tcp_read_req:
content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from
the complete log..

A             kamailo                 B

----INVITE-->- --------------------INVITE

180 Ringing--<---------------180 Ringing
200 OK-----------<---------------200 OK 0
200 OK-----------<---------------200 OK 500ms
200 OK-----------<---------------200 OK 1s
200 OK-----------<---------------200 OK 2s

< From here it is not passing to the application- script>

200 OK-----------<---------------200 OK 4s
200 OK-----------<---------------200 OK 4s



  28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 0
  282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102
<sip:5616990102 at 192.168.243.161:5060;transport=tcp>
  331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102
<sip:5616990102 at 192.168.243.161:5060;transport=tcp>
  398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset
  406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102
<sip:5616990102 at 192.168.243.161:5060;transport=tcp>
  481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102
<sip:5616990102 at 192.168.243.161:5060;transport=tcp>
  556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102
<sip:5616990102 at 192.168.243.161:5060;transport=tcp>
  631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG:
<core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
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From klaus.mailinglists at pernau.at  Tue Jun 14 12:31:30 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 14 Jun 2011 12:31:30 +0200
Subject: [SR-Users] RTP proxy
In-Reply-To: <BANLkTimtyz88ehhj6finEcn42ZobkxtB4w@mail.gmail.com>
References: <BANLkTimtyz88ehhj6finEcn42ZobkxtB4w@mail.gmail.com>
Message-ID: <4DF73882.8090401@pernau.at>

Are you sure that force_rtp_proxy is not called?


Instead of
> if (nat_uac_test("8")) {
>                     *    force_rtp_proxy();*
>                 } else {
>                         force_rtp_proxy();
>                 }

try:

xlog("L_ERR","route(4): method = $rm");
if (is_method("BYE|CANCEL")) {
  xlog("L_ERR","route(4): cancel or bye -> unforce_rtp_proxy ...");
  unforce_rtp_proxy();
  xlog("L_ERR","route(4): cancel or bye -> unforce_rtp_proxy ... done");
} else if (is_method("INVITE")){
  xlog("L_ERR","route(4): INVITE -> force_rtp_proxy ...");
  force_rtp_proxy();
  xlog("L_ERR","route(4): INVITE -> force_rtp_proxy ...");
}

also you can try to increase debug level and make sure that rtp proxy is
running.

You can alos change rtpproxy to listen on a UDP socket and reconfigure
kamailio to connect to rtpproxy via UDP transport - then you can watch
communication between kamailio and rtpproxy using a packet sniffer on
the loopback interface.

regards
klaus


From spinov_evgeniy at intalisan.com  Tue Jun 14 13:06:07 2011
From: spinov_evgeniy at intalisan.com (Evgeniy Spinov)
Date: Tue, 14 Jun 2011 12:06:07 +0100
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <4DF5E296.6070607@gmail.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com> <13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
	<4DF5E296.6070607@gmail.com>
Message-ID: <1308049573.1569.6.camel@localhost.localdomain>

On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:

Ok, still unsolved problem.

I've changed failover part of the script to use ds_next_dst(). It does
what it should and signalization works fine - calls are being connected.
However I've got no audio and message in logs:

7(9917) ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply
from rtp proxy

I've added force_rtp_proxy() after ds_next_dst() and message
disappeared, however still no audio on the failover call. 

RTP Proxy runs on the same server and is fine for not failover calls.
Success route script executes force_rtp_proxy() and fix_nated_contact()
in curtain conditions and that's it.

Am I missing something in my logic?

Thank you.

> 
> On 6/13/11 12:03 PM, Evgeniy Spinov wrote:
> > On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
> >> Hello,
> >>
> >> On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
> >>> Hello.
> >>>
> >>>
> >>> DCM>   Hello,
> >>>
> >>> DCM>   to understand the scenario:
> >>> DCM>   - first branch has destination uri ($du) set
> >>>
> >>> Exactly.
> >>>
> >>> DCM>   - it failed and gets to failure route where you call ds_next_domain()
> >>> DCM>   and $du s still the one from first branch?
> >>>
> >>> Exactly. $du is being set by ds_select_dst() before t_relay() before
> >>> any branch in main loop. In failure route after ds_next_domain()
> >>> variable $du remains with old IP ( previous asterisk node ), while $rd
> >>> variable is updated.
> >> is any particular reason to use ds_next_domain() after ds_select_dst()?
> >> ds_next_domain() is the pair for ds_select_domain, in older versions it
> >> happened that $du was reset automatically internally even by
> >> ds_next_domain() due to execution of a core function which was no longer
> >> needed in 3.1.
> > I do not mind using ds_select_dst() however it seems to me, that in
> > following scenario it will give me a false result:
> > 1. Call 1 is coming
> > 2. Routed to GW 1
> > 3. Call 2 is coming
> > 4. Routed to GW 2
> > 5. Call 1 timed out to GW 1 and call is going to failure route, where
> > ds_select_dst() is being called.
> > I guess ds_select_dst in this case will give GW 1 again in case of
> > round-robin fashion.
> >
> > Also, there is no end of gateways, as ds_select_dst() will always return
> > one of the gateways, while when no gateways are alive - appropriate
> > reaction should happen.
> >
> > That's why ds_next_domain() is more suitable, as performing all the
> > required actions without magic. Performed at least.
> ds_select_dst()/ds_next_dst() and ds_select_domain()/ds_next_domain() 
> consume the already-tried destinations -- in any of the cases, none of 
> gateways will be tried two times.
> 
> 
> ds_*_dst() sets $du and $ds_*_domain() sets $ru -- this is the 
> difference between them.
> 
> When you do first ds_select_dst() you set $du and when then you do 
> $ds_next_domain() you set $ru -- if you would do $ds_next_dst() then $du 
> will be set with the new gateway address.
> 
> >> A quick solution if you need ds_select_dst() in combination with
> >> ds_next_domain() is to call resetdsturi() in failure route. I will think
> >> about and see if it is consistent to call it internally.
> > Works for me.
> >
> >>> DCM>   What do you mean that "not any of the nodes receive the packet ..."?
> >>>
> >>> Kamailio is standing before asterisks and relays packets to them. In
> >>> case, when there is no ds_next_domain() - packet is routed correctly
> >>> to the curtain node, defined by ds_select_dst(). While after
> >>> ds_next_domain() none of the 2 nodes receives the packet and call
> >>> terminates by the caller with "Request timeout". Looks like t_relay()
> >>> sends packet somewhere to blackhole.
> >> Can you look at sip trace with ngrep to see where they are sent?
> >> Probably to the first destination that failed.
> > Yes. It sends packet to previous gateway, while in INVITE another IP
> > mentioned ( look at 1 and 2 strings ):
> >
> > 13:59:36.533705 IP KAMAILIO_IP.sip>  GW1_IP.sip: UDP, length 1020
> > E....H.. at .4..(...(.........UINVITE sip:*44 at GW2_IP:5060 SIP/2.0
> > Record-Route:
> > <sip:KAMAILIO_IP;lr=on;ftag=ustdc;did=c4e.a81165f4;nat=yes>
> > Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bKccae.c54a1682.1
> > Max-Forwards: 69
> > To:<sip:*44 at KAMAILIO_IP>
> > From: "2_101"<sip:2_101 at KAMAILIO_IP>;tag=ustdc
> > Call-ID: yuwofwddoaqozwh at localhost.localdomain
> > CSeq: 527 INVITE
> > Contact:<sip:2_101 at UAC_PUBLIC_IP:5061>
> > Content-Type: application/sdp
> > Allow:
> > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> > Supported: replaces,norefersub,100rel
> > User-Agent: Twinkle/1.4.2
> > Content-Length: 330
> >
> >
> >
> > Looks like a bug.
> 
> This is not a bug in sip routing as long as $du is set, then it is the 
> field used for sip routing no matter you have in $ru -- $du is the 
> internal outbound proxy address overwriting request-uri address.
> 
> Maybe will be safe to introduce resetdsturi() in case of 
> ds_next_domain() to be safe in combinations like 
> ds_select_dst()+ds_next_domain().
> 
> Cheers,
> Daniel
> 
> 
> 




From ibc at aliax.net  Tue Jun 14 14:27:53 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Tue, 14 Jun 2011 14:27:53 +0200
Subject: [SR-Users] Use of mtree module having multiple trees in the
	same table
In-Reply-To: <BANLkTik+dR8iBdrr1vtWfjzdZXpsujWzWw@mail.gmail.com>
References: <BANLkTik+dR8iBdrr1vtWfjzdZXpsujWzWw@mail.gmail.com>
Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww@mail.gmail.com>

2011/6/13 jose luis millan <doctore96 at gmail.com>:
> I wonder if anyone could help me understanding how can I use this
> module in a scenario where multiple mtrees are defined behind the same
> database table.
>
> I can't guess how to speficy to 'mt_match' the mtree I want to match
> against, since the 'mtree' module parameter defines a specific memory
> tree based on a specific table, but not based on a specific 'tname'
> value.

Indeed we have a table with these colums:

- prefix:    string to identify number destination. This is the regex column.
- id_zone:  integer, points to other table. This is the value column.
- scope:    integer, can be 0 or 1.

So wee need two instances of mtree:
1) those rows with scope 0.
2) those rows with scope 1.

Is it possible with current mtree module? or maybe we'd do better by
creating two views of the table (one for entries with scope 0 and
another for entries with scope 1)?

Thanks a lot.

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


From realjijo at gmail.com  Tue Jun 14 15:59:54 2011
From: realjijo at gmail.com (Jijo)
Date: Tue, 14 Jun 2011 09:59:54 -0400
Subject: [SR-Users] Fwd: 200OK Retransmission Error
In-Reply-To: <BANLkTi=Sj0x0AMNzWJGLf-76aZECL-wQZQ@mail.gmail.com>
References: <BANLkTi=6ngJ=FXkGnBj6P0C9001UgL6ymQ@mail.gmail.com>
	<BANLkTi=Sj0x0AMNzWJGLf-76aZECL-wQZQ@mail.gmail.com>
Message-ID: <BANLkTimwG-NeQYXg+9GQvw8VUunu5FMmLQ@mail.gmail.com>

Hi All,

In an INVITE transaction, no ACK from A party and B party continue
retransmission of 200 OK. I have observed that kamailio is not passing the
retransmitted 200 OK message from B party to the Application(script) after 3
seconds. Instead kamailio directly sending to A party without passing to the
script.

I see in the logs that kamailio is cleaning up the transaction after 3
seconds..look for the string "DEBUG: t_lookup_request: no transaction
found"  before this log it is cleaning up the timer in the log "DEBUG:
cleanup_uac_timers: RETR/FR timers reset"

What does it mean by that??
How can we make the kamailio to pass the response to the application?

Please let me know your comments.. Here is the flow. I cannot send the
complete logs which is quite big. so i grep'd using the stirng
"tcp_read_req:
content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from
the complete log..

A             kamailo                 B

----INVITE-->- --------------------INVITE

180 Ringing--<---------------180 Ringing
200 OK-----------<---------------200 OK 0
200 OK-----------<---------------200 OK 500ms
200 OK-----------<---------------200 OK 1s
200 OK-----------<---------------200 OK 2s

< From here it is not passing to the application- script>

200 OK-----------<---------------200 OK 4s
200 OK-----------<---------------200 OK 4s



  28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 0
  282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip:
5616990102 at 192.168.243.161:5060;transport=tcp>
  331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset
  406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG:
<core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
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From doctore96 at gmail.com  Tue Jun 14 16:20:30 2011
From: doctore96 at gmail.com (jose luis millan)
Date: Tue, 14 Jun 2011 16:20:30 +0200
Subject: [SR-Users] Use of mtree module having multiple trees in the
	same table
In-Reply-To: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww@mail.gmail.com>
References: <BANLkTik+dR8iBdrr1vtWfjzdZXpsujWzWw@mail.gmail.com>
	<BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww@mail.gmail.com>
Message-ID: <BANLkTimOGPvt6y6DCQ2_g-T7rqCb9nDtXA@mail.gmail.com>

Hi,

In order to have multiple trees in a single table,  function
'mt_match' has to be called with the desired 'tname'  value as the
first argument.

So, the first parameter in the 'mt_match' function refers to a value
of 'tname' in case this parameter is used, or to an explicitly created
mtree (defined via the mtree par?meter).

Thanks



On Tue, Jun 14, 2011 at 2:27 PM, I?aki Baz Castillo <ibc at aliax.net> wrote:
> 2011/6/13 jose luis millan <doctore96 at gmail.com>:
>> I wonder if anyone could help me understanding how can I use this
>> module in a scenario where multiple mtrees are defined behind the same
>> database table.
>>
>> I can't guess how to speficy to 'mt_match' the mtree I want to match
>> against, since the 'mtree' module parameter defines a specific memory
>> tree based on a specific table, but not based on a specific 'tname'
>> value.
>
> Indeed we have a table with these colums:
>
> - prefix: ? ?string to identify number destination. This is the regex column.
> - id_zone: ?integer, points to other table. This is the value column.
> - scope: ? ?integer, can be 0 or 1.
>
> So wee need two instances of mtree:
> 1) those rows with scope 0.
> 2) those rows with scope 1.
>
> Is it possible with current mtree module? or maybe we'd do better by
> creating two views of the table (one for entries with scope 0 and
> another for entries with scope 1)?
>
> Thanks a lot.
>
> --
> 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 lucasaa at gmail.com  Wed Jun 15 00:52:51 2011
From: lucasaa at gmail.com (Lucas Alvarez)
Date: Tue, 14 Jun 2011 19:52:51 -0300
Subject: [SR-Users] Modifying sip packet
Message-ID: <BANLkTinzju6_4xMTRiQ4s+TdpgkhM3rjWg@mail.gmail.com>

Hi, I want to modify  the following sip packet, I want to change the SDP IP
address by $di, which rule could filter this packet? Something like: [ if
method("OK") and $di is RFC1918 ] ??? And with which function could I change
the Ip address in the SDP content? In this case I would like to change
172.16.99.2
by 190.244.125.41.
Thanks in advance,

Lucas

T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP]
 SIP/2.0 200 OK.
Via: SIP/2.0/UDP
172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060.
Via: SIP/2.0/TCP 192.168.15.231:1952
;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055.
Record-Route: <sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e>.
Record-Route: <sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e>.
From: "Lucas"<sip:1104 at 67.152.18.231>;tag=b11c6a4e.
To: <sip:1101 at 67.152.18.231>;tag=as5356a251.
Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM..
CSeq: 2 INVITE.
Server: Asterisk PBX 1.8.4.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Contact: <sip:1101 at 172.16.99.2:5080>.
Remote-Party-ID: "Gus Office One" <sip:1101 at 67.152.18.231
>;party=called;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 305.
.
v=0.
o=root 1124623741 1124623741 IN IP4 172.16.99.2.
s=Asterisk PBX 1.8.4.1.
c=IN IP4 172.16.99.2.
t=0 0.
m=audio 13640 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=ptime:20.
a=sendrecv.
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From osas at voipembedded.com  Wed Jun 15 00:58:35 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Tue, 14 Jun 2011 18:58:35 -0400
Subject: [SR-Users] Modifying sip packet
In-Reply-To: <BANLkTinzju6_4xMTRiQ4s+TdpgkhM3rjWg@mail.gmail.com>
References: <BANLkTinzju6_4xMTRiQ4s+TdpgkhM3rjWg@mail.gmail.com>
Message-ID: <BANLkTin_Sz-rgiu762_=fcz+ZzZWLLGsoA@mail.gmail.com>

Take a look at the nathelper module:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/nathelper.html#id2592146


Regards,
Ovidiu Sas

On Tue, Jun 14, 2011 at 6:52 PM, Lucas Alvarez <lucasaa at gmail.com> wrote:
> Hi, I want to modify ?the following sip packet, I want to change the SDP IP
> address by $di, which rule could filter this packet? Something like: [ if
> method("OK") and $di is RFC1918 ] ??? And with which function could I change
> the Ip address in the SDP content? In this case I would like to
> change?172.16.99.2 by?190.244.125.41.
> Thanks in advance,
> Lucas
> T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP]
> ?SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060.
> Via: SIP/2.0/TCP
> 192.168.15.231:1952;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055.
> Record-Route: <sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e>.
> Record-Route: <sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e>.
> From: "Lucas"<sip:1104 at 67.152.18.231>;tag=b11c6a4e.
> To: <sip:1101 at 67.152.18.231>;tag=as5356a251.
> Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM..
> CSeq: 2 INVITE.
> Server: Asterisk PBX 1.8.4.1.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH.
> Supported: replaces, timer.
> Contact: <sip:1101 at 172.16.99.2:5080>.
> Remote-Party-ID: "Gus Office One"
> <sip:1101 at 67.152.18.231>;party=called;privacy=off;screen=no.
> Content-Type: application/sdp.
> Content-Length: 305.
> .
> v=0.
> o=root 1124623741 1124623741 IN IP4 172.16.99.2.
> s=Asterisk PBX 1.8.4.1.
> c=IN IP4 172.16.99.2.
> t=0 0.
> m=audio 13640 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=ptime:20.
> a=sendrecv.
>
> _______________________________________________
> 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 omar at 321communications.com  Wed Jun 15 02:04:08 2011
From: omar at 321communications.com (Omar)
Date: Tue, 14 Jun 2011 20:04:08 -0400
Subject: [SR-Users] 302 rewrite ruri based on contact
In-Reply-To: <BANLkTikiuah0K-UUvbKHNw5fCaRS=aBkKg@mail.gmail.com>
References: <AB1828DB-A9EB-4B10-9894-F87AFA3DBD6E@321communications.com>
	<BANLkTinUL-UopvEMBJROisVxybt2Hq7xPg@mail.gmail.com>
	<1DDC855A-2737-4784-9B53-37352BAC3683@321communications.com>
	<BANLkTikiuah0K-UUvbKHNw5fCaRS=aBkKg@mail.gmail.com>
Message-ID: <BE43E418-3308-4F00-B413-EF42E3088C87@321communications.com>

what other option can i use for the following scenario

In Kamailio 1.4.3

I have the following scenario:

$var(a) = 0
$var(b) = 0
$var(a) = $(tU{s.len});
$var(b) =  $var(a) - 5;

xlog(" $var(b)");
$var(a) is 8
the result $var(b)  is always the same as $var(a)

I used cfgutils to initialize the variables as integer

and nothing works.

if i use AVP

$avp(i:a) = $(tU{s.len}); i got an errors showing the a name is not correct, and does not contain a number !

s.len is a number right?


regards

O

On Jun 8, 2011, at 9:15 AM, Ovidiu Sas wrote:

> $var are script variables (read more here):
> http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#script_variables
> Best thing to do is to initialize the variable every time before using it.
> The variable is not transaction or call persistent (it's scope is in
> context of the message being processed).
> 
> $avp may be transaction persistent (enable it in tm):
> http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#avps
> http://www.kamailio.org/docs/modules/1.4.x/tm.html#id2530425
> 
> 
> Regards,
> Ovidiu Sas
> 
> 
> On Wed, Jun 8, 2011 at 7:10 AM, Omar <omar at 321communications.com> wrote:
>> Do i have to clear the variable in $var everytime a call ends a Bye or cancels, or the variable will clear it up automatically in Kamailio1.4.3? i think this acts as a vector, but i have some doubt regarding
>> 
>> $avp i think uses shared memory, then i have to take care to reserve enough memory for it in the shared memory right?
>> 
>> 
>> thanks
>> 
>> O
>> 
>> 
>> On May 24, 2011, at 6:58 PM, Ovidiu Sas wrote:
>> 
>>> Here's an example on how to perform forking on failure on 1.4:
>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4/examples/serial_183.cfg
>>> 
>>> 
>>> Regards,
>>> Ovidiu Sas
>>> 
>>> On Tue, May 24, 2011 at 4:30 PM, Omar <omar at 321communications.com> wrote:
>>>> 
>>>> I need to rewrite the destination after I receive 302
>>>> I have this logic
>>>> 
>>>> Onreply
>>>> Loads in avp the information needed
>>>> 
>>>> I try sending to failure route and to a route in both cases the $ru become <null>
>>>> 
>>>> In the route and failureroute
>>>> I did try rewritehost, rewriteuri, $ru= number+ host
>>>> And append the route
>>>> But $ru is always null.
>>>> Inside the route or failure route
>>>> 
>>>> If there are no 302 nor the onreply involved everything works fine
>>>> 
>>>> Any suggestion to rewrite the ruri after a 302 is welcome
>>>> 
>>>> I am still using Kamailo1.4.3
>>>> 
>>>> 
>>>> Omar
>>>> _______________________________________________
>>>> 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
>>> 
>> 
>> 
>> _______________________________________________
>> 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 realjijo at gmail.com  Wed Jun 15 06:16:28 2011
From: realjijo at gmail.com (Jijo)
Date: Wed, 15 Jun 2011 00:16:28 -0400
Subject: [SR-Users] 200 OK Retransmission Error
Message-ID: <BANLkTim_TaNti=H1cJ=bGj=7a4-DWZa8=w@mail.gmail.com>

Hi All,

In an INVITE transaction, no ACK from A party and B party continue
retransmission of 200 OK. I have observed that kamailio is not passing the
retransmitted 200 OK message from B party to the Application(script) after 3
seconds. Instead kamailio directly sending to A party without passing to the
script.

I see in the logs that kamailio is cleaning up the transaction after 3
seconds..look for the string "DEBUG: t_lookup_request: no transaction
found"  before this log it is cleaning up the timer in the log "DEBUG:
cleanup_uac_timers: RETR/FR timers reset"

What does it mean by that??
How can we make the kamailio to pass the response to the application?

Please let me know your comments.. Here is the flow. I cannot send the
complete logs which is quite big. so i grep'd using the stirng
"tcp_read_req:
content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from
the complete log..

A             kamailo                 B

----INVITE-->- --------------------INVITE

180 Ringing--<---------------180 Ringing
200 OK-----------<---------------200 OK 0
200 OK-----------<---------------200 OK 500ms
200 OK-----------<---------------200 OK 1s
200 OK-----------<---------------200 OK 2s

< From here it is not passing to the application- script>

200 OK-----------<---------------200 OK 4s
200 OK-----------<---------------200 OK 4s



  28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 0
  282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip:
5616990102 at 192.168.243.161:5060;transport=tcp>
  331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset
  406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
  581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>:
CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From=
sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060 RD=
<null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102
@192.168.243.161:5060;transport=tcp>
  631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
  1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core>
[tcp_read.c:933]: tcp_read_req: content-length= 191
  1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG:
<core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
sip:5616990102 at 192.168.243.151:5060]
  1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core>
[parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
  1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
0
  1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
  1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
[t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
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From miconda at gmail.com  Wed Jun 15 09:47:09 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Jun 2011 09:47:09 +0200
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <1308049573.1569.6.camel@localhost.localdomain>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com>
	<13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
	<4DF5E296.6070607@gmail.com>
	<1308049573.1569.6.camel@localhost.localdomain>
Message-ID: <4DF8637D.3080306@gmail.com>



On 6/14/11 1:06 PM, Evgeniy Spinov wrote:
> On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:
>
> Ok, still unsolved problem.
>
> I've changed failover part of the script to use ds_next_dst(). It does
> what it should and signalization works fine - calls are being connected.
> However I've got no audio and message in logs:
>
> 7(9917) ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply
> from rtp proxy
>
> I've added force_rtp_proxy() after ds_next_dst() and message
> disappeared, however still no audio on the failover call.
>
> RTP Proxy runs on the same server and is fine for not failover calls.
> Success route script executes force_rtp_proxy() and fix_nated_contact()
> in curtain conditions and that's it.
>
> Am I missing something in my logic?
Try to do the rtpproxy thing in branch route -- anyhow this has nothing 
to with dispatcher.

Cheers,
Daniel

>
> Thank you.
>
>> On 6/13/11 12:03 PM, Evgeniy Spinov wrote:
>>> On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
>>>> Hello,
>>>>
>>>> On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
>>>>> Hello.
>>>>>
>>>>>
>>>>> DCM>    Hello,
>>>>>
>>>>> DCM>    to understand the scenario:
>>>>> DCM>    - first branch has destination uri ($du) set
>>>>>
>>>>> Exactly.
>>>>>
>>>>> DCM>    - it failed and gets to failure route where you call ds_next_domain()
>>>>> DCM>    and $du s still the one from first branch?
>>>>>
>>>>> Exactly. $du is being set by ds_select_dst() before t_relay() before
>>>>> any branch in main loop. In failure route after ds_next_domain()
>>>>> variable $du remains with old IP ( previous asterisk node ), while $rd
>>>>> variable is updated.
>>>> is any particular reason to use ds_next_domain() after ds_select_dst()?
>>>> ds_next_domain() is the pair for ds_select_domain, in older versions it
>>>> happened that $du was reset automatically internally even by
>>>> ds_next_domain() due to execution of a core function which was no longer
>>>> needed in 3.1.
>>> I do not mind using ds_select_dst() however it seems to me, that in
>>> following scenario it will give me a false result:
>>> 1. Call 1 is coming
>>> 2. Routed to GW 1
>>> 3. Call 2 is coming
>>> 4. Routed to GW 2
>>> 5. Call 1 timed out to GW 1 and call is going to failure route, where
>>> ds_select_dst() is being called.
>>> I guess ds_select_dst in this case will give GW 1 again in case of
>>> round-robin fashion.
>>>
>>> Also, there is no end of gateways, as ds_select_dst() will always return
>>> one of the gateways, while when no gateways are alive - appropriate
>>> reaction should happen.
>>>
>>> That's why ds_next_domain() is more suitable, as performing all the
>>> required actions without magic. Performed at least.
>> ds_select_dst()/ds_next_dst() and ds_select_domain()/ds_next_domain()
>> consume the already-tried destinations -- in any of the cases, none of
>> gateways will be tried two times.
>>
>>
>> ds_*_dst() sets $du and $ds_*_domain() sets $ru -- this is the
>> difference between them.
>>
>> When you do first ds_select_dst() you set $du and when then you do
>> $ds_next_domain() you set $ru -- if you would do $ds_next_dst() then $du
>> will be set with the new gateway address.
>>
>>>> A quick solution if you need ds_select_dst() in combination with
>>>> ds_next_domain() is to call resetdsturi() in failure route. I will think
>>>> about and see if it is consistent to call it internally.
>>> Works for me.
>>>
>>>>> DCM>    What do you mean that "not any of the nodes receive the packet ..."?
>>>>>
>>>>> Kamailio is standing before asterisks and relays packets to them. In
>>>>> case, when there is no ds_next_domain() - packet is routed correctly
>>>>> to the curtain node, defined by ds_select_dst(). While after
>>>>> ds_next_domain() none of the 2 nodes receives the packet and call
>>>>> terminates by the caller with "Request timeout". Looks like t_relay()
>>>>> sends packet somewhere to blackhole.
>>>> Can you look at sip trace with ngrep to see where they are sent?
>>>> Probably to the first destination that failed.
>>> Yes. It sends packet to previous gateway, while in INVITE another IP
>>> mentioned ( look at 1 and 2 strings ):
>>>
>>> 13:59:36.533705 IP KAMAILIO_IP.sip>   GW1_IP.sip: UDP, length 1020
>>> E....H.. at .4..(...(.........UINVITE sip:*44 at GW2_IP:5060 SIP/2.0
>>> Record-Route:
>>> <sip:KAMAILIO_IP;lr=on;ftag=ustdc;did=c4e.a81165f4;nat=yes>
>>> Via: SIP/2.0/UDP KAMAILIO_IP;branch=z9hG4bKccae.c54a1682.1
>>> Max-Forwards: 69
>>> To:<sip:*44 at KAMAILIO_IP>
>>> From: "2_101"<sip:2_101 at KAMAILIO_IP>;tag=ustdc
>>> Call-ID: yuwofwddoaqozwh at localhost.localdomain
>>> CSeq: 527 INVITE
>>> Contact:<sip:2_101 at UAC_PUBLIC_IP:5061>
>>> Content-Type: application/sdp
>>> Allow:
>>> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
>>> Supported: replaces,norefersub,100rel
>>> User-Agent: Twinkle/1.4.2
>>> Content-Length: 330
>>>
>>>
>>>
>>> Looks like a bug.
>> This is not a bug in sip routing as long as $du is set, then it is the
>> field used for sip routing no matter you have in $ru -- $du is the
>> internal outbound proxy address overwriting request-uri address.
>>
>> Maybe will be safe to introduce resetdsturi() in case of
>> ds_next_domain() to be safe in combinations like
>> ds_select_dst()+ds_next_domain().
>>
>> Cheers,
>> Daniel
>>
>>
>>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Wed Jun 15 09:56:26 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Jun 2011 09:56:26 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <0847B1C4-F998-4471-9870-52E247928CBD@gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
	<BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
	<BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>
	<4DF5E0A1.1080605@gmail.com>
	<BANLkTinqSkPH=JtACxqaoEQU05eZk35oCg@mail.gmail.com>
	<0847B1C4-F998-4471-9870-52E247928CBD@gmail.com>
Message-ID: <4DF865AA.2030509@gmail.com>

One more question here, since I remembered about it -- have you played 
with dns-based load balancing done by proxy so far? This was new for 
kamailio in 3.x...

Thanks,
Daniel

On 6/13/11 3:55 PM, Daniel-Constantine Mierla wrote:
>
> On Jun 13, 2011, at 2:48 PM, I?aki Baz Castillo<ibc at aliax.net>  wrote:
>
>> 2011/6/13 Daniel-Constantin Mierla<miconda at gmail.com>:
>>> I saw Andrei jumped in and added case insensitive comparison of naptr flags
>>> -- just to conclude this discussion, is it working on now?
>> Hi Daniel, I've tested it right *now* :)
>> Yes, it works.
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Wed Jun 15 09:52:14 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Jun 2011 09:52:14 +0200
Subject: [SR-Users] 200 OK Retransmission Error
In-Reply-To: <BANLkTim_TaNti=H1cJ=bGj=7a4-DWZa8=w@mail.gmail.com>
References: <BANLkTim_TaNti=H1cJ=bGj=7a4-DWZa8=w@mail.gmail.com>
Message-ID: <4DF864AE.3010905@gmail.com>

Hello,

On 6/15/11 6:16 AM, Jijo wrote:
>
>
> Hi All,
>
> In an INVITE transaction, no ACK from A party and B party continue 
> retransmission of 200 OK. I have observed that kamailio is not passing 
> the retransmitted 200 OK message from B party to the 
> Application(script) after 3 seconds. Instead kamailio directly sending 
> to A party without passing to the script.
what do you mean "without passing to the script"?

Note that a transaction is completed when a final reply is forwarded 
back to the sender of request. Then it is kept fore a short time in 
memory, if you want to increase that interval, see this parameter in tm:

http://kamailio.org/docs/modules/stable/modules/tm.html#wt_timer

Cheers,
Daniel

>
> I see in the logs that kamailio is cleaning up the transaction after 3 
> seconds..look for the string "DEBUG: t_lookup_request: no transaction 
> found"  before this log it is cleaning up the timer in the log "DEBUG: 
> cleanup_uac_timers: RETR/FR timers reset"
>
> What does it mean by that??
> How can we make the kamailio to pass the response to the application?
>
> Please let me know your comments.. Here is the flow. I cannot send the 
> complete logs which is quite big. so i grep'd using the stirng 
> "tcp_read_req: 
> content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" 
> from the complete log..
>
> A             kamailo                 B
>
> ----INVITE-->- --------------------INVITE
>
> 180 Ringing--<---------------180 Ringing
> 200 OK-----------<---------------200 OK 0
> 200 OK-----------<---------------200 OK 500ms
> 200 OK-----------<---------------200 OK 1s
> 200 OK-----------<---------------200 OK 2s
>
> < From here it is not passing to the application- script>
>
> 200 OK-----------<---------------200 OK 4s
> 200 OK-----------<---------------200 OK 4s
>
>
>
>   28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 0
>   282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched 
> (T=0xb0f62e50)!
>   292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: 
> <script>: CI=1-8028 at 192.168.243.150 <mailto:1-8028 at 192.168.243.150> 
> -RR11 - Entry M=INVITE RURI=<null> From=sip:5616990103 at 192.168.243.150 
> <mailto:sip%3A5616990103 at 192.168.243.150> 
> To=sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060> RD= <null> 
> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 
> <tel:5616990102> <sip:5616990102 
> <tel:5616990102>@192.168.243.161:5060;transport=tcp>
>   331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched 
> (T=0xb0f62e50)!
>   356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: 
> <script>: CI=1-8028 at 192.168.243.150 <mailto:1-8028 at 192.168.243.150> 
> -RR11 - Entry M=INVITE RURI=<null> From=sip:5616990103 at 192.168.243.150 
> <mailto:sip%3A5616990103 at 192.168.243.150> 
> To=sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060> RD= <null> 
> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <tel:5616990102> 
> <sip:5616990102 <tel:5616990102>@192.168.243.161:5060;transport=tcp>
>   398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset
>   406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched 
> (T=0xb0f62e50)!
>   431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: 
> <script>: CI=1-8028 at 192.168.243.150 <mailto:1-8028 at 192.168.243.150> 
> -RR11 - Entry M=INVITE RURI=<null> From=sip:5616990103 at 192.168.243.150 
> <mailto:sip%3A5616990103 at 192.168.243.150> 
> To=sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060> RD= <null> 
> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <tel:5616990102> 
> <sip:5616990102 <tel:5616990102>@192.168.243.161:5060;transport=tcp>
>   481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched 
> (T=0xb0f62e50)!
>   506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: 
> <script>: CI=1-8028 at 192.168.243.150 <mailto:1-8028 at 192.168.243.150> 
> -RR11 - Entry M=INVITE RURI=<null> From=sip:5616990103 at 192.168.243.150 
> <mailto:sip%3A5616990103 at 192.168.243.150> 
> To=sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060> RD= <null> 
> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <tel:5616990102> 
> <sip:5616990102 <tel:5616990102>@192.168.243.161:5060;transport=tcp>
>   556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched 
> (T=0xb0f62e50)!
>   581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: 
> <script>: CI=1-8028 at 192.168.243.150 <mailto:1-8028 at 192.168.243.150> 
> -RR11 - Entry M=INVITE RURI=<null> From=sip:5616990103 at 192.168.243.150 
> <mailto:sip%3A5616990103 at 192.168.243.150> 
> To=sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060> RD= <null> 
> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <tel:5616990102> 
> <sip:5616990102 <tel:5616990102>@192.168.243.161:5060;transport=tcp>
>   631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>   693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>   755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>   817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>   879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>   941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>   1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; 
> uri=[sip:5616990102 at 192.168.243.151:5060 
> <http://sip:5616990102 at 192.168.243.151:5060>]
>   1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: 
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 
> 418362480 branch 0
>   1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction 
> exists
>   1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: 
> tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a 
> transaction
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From ibc at aliax.net  Wed Jun 15 10:11:06 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Wed, 15 Jun 2011 10:11:06 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <4DF865AA.2030509@gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<4DF20895.6050606@pernau.at>
	<BANLkTimSziMfqOePMdJ0hG8cUrsyWEVERA@mail.gmail.com>
	<BANLkTik_XNAuZpGtGnNrZ8V-Q5tQqTo2mQ@mail.gmail.com>
	<4DF218A9.7050205@pernau.at>
	<BANLkTimwOtmsKHCmvcpxJPg=Fvf-Di_SWw@mail.gmail.com>
	<BANLkTinwNLNCj-y7XeJR+UANVkk4=RAkig@mail.gmail.com>
	<4DF27937.8000803@gmail.com>
	<BANLkTim1VSKehTBpwskFkQvqCX+mi7S8yw@mail.gmail.com>
	<4DF28B5D.9010404@kamailio.org>
	<BANLkTinKKSG0fPGCwnu786bswyqhOPMq+w@mail.gmail.com>
	<BANLkTi=2xCArRCE_HXkXcoU6aPjqUZ_Y9w@mail.gmail.com>
	<4DF5E0A1.1080605@gmail.com>
	<BANLkTinqSkPH=JtACxqaoEQU05eZk35oCg@mail.gmail.com>
	<0847B1C4-F998-4471-9870-52E247928CBD@gmail.com>
	<4DF865AA.2030509@gmail.com>
Message-ID: <BANLkTinWhr_PJrCEOGq33xKsRhdL8eHghQ@mail.gmail.com>

2011/6/15 Daniel-Constantin Mierla <miconda at gmail.com>:
> One more question here, since I remembered about it -- have you played with
> dns-based load balancing done by proxy so far? This was new for kamailio in
> 3.x...

I started playing a bit and got good results, but want to make further tests.

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


From spinov_evgeniy at intalisan.com  Wed Jun 15 10:50:42 2011
From: spinov_evgeniy at intalisan.com (Evgeniy Spinov)
Date: Wed, 15 Jun 2011 09:50:42 +0100
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <4DF8637D.3080306@gmail.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com> <13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
	<4DF5E296.6070607@gmail.com>
	<1308049573.1569.6.camel@localhost.localdomain>
	<4DF8637D.3080306@gmail.com>
Message-ID: <1308127843.1551.8.camel@localhost.localdomain>

On Wed, 2011-06-15 at 09:47 +0200, Daniel-Constantin Mierla wrote:
> 
> On 6/14/11 1:06 PM, Evgeniy Spinov wrote:
> > On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:
> >
> > Ok, still unsolved problem.
> >
> > I've changed failover part of the script to use ds_next_dst(). It does
> > what it should and signalization works fine - calls are being connected.
> > However I've got no audio and message in logs:
> >
> > 7(9917) ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply
> > from rtp proxy
> >
> > I've added force_rtp_proxy() after ds_next_dst() and message
> > disappeared, however still no audio on the failover call.
> >
> > RTP Proxy runs on the same server and is fine for not failover calls.
> > Success route script executes force_rtp_proxy() and fix_nated_contact()
> > in curtain conditions and that's it.
> >
> > Am I missing something in my logic?
> Try to do the rtpproxy thing in branch route -- anyhow this has nothing 
> to with dispatcher.
> 
> Cheers,
> Daniel
> 

It's done on the branch route, however it seems that branch route is not
being executed when the failure branch route is being ran. Anyhow, when
the call is not passing through failure route and ds_next_dst() it is
fine, no errors. When it goes there, error with rtp proxy in logs and no
audio. When I add force_rtp_proxy() on success of t_relay() in failure
route it gives no errors, but still no audio. 

I do not know where issue is related, and what is wrong, especially when
this scenario worked before.

Just to avoid misunderstanding, providing you a failure and a normal
branches of the script. Failure route:

        if (is_method("INVITE")
                        && (isbflagset("6") || isflagset(5))) {
                unforce_rtp_proxy();
        }

	... ( checks for different errors and messages output is skipped, not
related to this issue anyway )

            if( ds_next_dst() )
            {
                xlog( "L_NOTICE", "[$Tf] $ci Next gateway $fU -> $tU via
$du\n" );
                if( !t_relay() )
                {
                    xlog( "L_INFO", "[$Tf] $ci -- ERROR - Can not
t_relay()\n" );
                    return;
                };
                return;
            }
            else
            {
                     xlog( "L_INFO", "[$Tf] $ci No more routes available
\n" );
                t_reply("503", "Service unavailable -- no more gateways"
);
                exit;
            };


And success route completely:

        if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") {
                force_rtp_proxy();
        }
        if (isbflagset("6")) {
                fix_nated_contact();
        }


Thank you.





From miconda at gmail.com  Wed Jun 15 14:24:10 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 15 Jun 2011 14:24:10 +0200
Subject: [SR-Users] Operations With Variables $var- does it need to load
 a module
In-Reply-To: <6C5E1099-D53A-4C6D-A36C-F0DC07D20C00@321communications.com>
References: <6C5E1099-D53A-4C6D-A36C-F0DC07D20C00@321communications.com>
Message-ID: <4DF8A46A.6010004@gmail.com>

Hello,

On 6/13/11 7:05 AM, Omar wrote:
> In Kamailio 1.4.3
>
> I have the following scenario:
>
> $var(a) = 0
> $var(b) = 0
> $var(a) = $(tU{s.len});
> $var(b) =  $var(a) - 5;
>
> xlog(" $var(b)");
>
> the result $var(b)  is always the same as $var(a)
>
> should i check if some specific module is loaded?
> the debugs does not show any error, and $var(a) and $var(b) are unique.
>
> Variables are working, but not the operations with the variables.
there is no module needed to be loaded for that version.

Can you add xlog() lines to print the values of variables for each step, 
including the $tU s.len transformation.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Wed Jun 15 14:32:24 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Wed, 15 Jun 2011 14:32:24 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=ijY08owMepH4K2yDrKQrNfQZzeQ@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<201106092026.02795.alex@speakup.nl>
	<BANLkTi=ijY08owMepH4K2yDrKQrNfQZzeQ@mail.gmail.com>
Message-ID: <BANLkTi=d1inRHxr2bQ+Rh4bZejg9sdFvgA@mail.gmail.com>

2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> 2011/6/9 Alex Hermann <alex at speakup.nl>:
>> The way I read rfc2915, there is no failover mechanism. The application pick
>> the first target that it supports and uses that. There is no mention of trying
>> other records afterwards. Matching/finding NAPTR records stops once the first
>> match is completed. All other records are discarded. From ?section 2:
>
> Hi Alex, two comments:
>
> 1) What you say does not explain my issue (as in any case TLS or TCP
> have more priority in my NAPTR's than UDP). In fact, my exact problem
> is that my Kamailio is not performing NAPTR query.

The problem is fixed now, it was an error related to the NAPTR service
flag (sip-router expected it to be "s" case-sensitive, while it should
also allow "S").


> 2) About your quoted text, indeed RFC 2915 says that. But that just
> means that, an application (in this case a SIP client) must take
> *first* the valid NAPTR record with better "order" (lowest value). But
> RFC 2915 is just about NAPTR, it can not mandate that applications or
> protocols cannot try other NAPTR records as failover mechanism. Any
> SIP client or proxy implementing NAPTR records does failover to the
> next NAPTR record in case the first one failed (at least those I've
> tested).

It seems that sip-router behaves exactly as you describe. This it, it
performs the NAPTR resolution, chooses the record with best 'order'
(for the supported transports) and just uses it (it would do SRV
failover and so, but would never try other NAPTR records with less
'order').

However I've seen other SIP stacks also performing failover by taking
next NAPTR records. But indeed, RFC 2915 does not mandate it, instead
it says MUST NOT. So you are right.


Regards.


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


From alex at speakup.nl  Wed Jun 15 15:09:21 2011
From: alex at speakup.nl (Alex Hermann)
Date: Wed, 15 Jun 2011 15:09:21 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTi=d1inRHxr2bQ+Rh4bZejg9sdFvgA@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTi=ijY08owMepH4K2yDrKQrNfQZzeQ@mail.gmail.com>
	<BANLkTi=d1inRHxr2bQ+Rh4bZejg9sdFvgA@mail.gmail.com>
Message-ID: <201106151509.22242.alex@speakup.nl>

On Wednesday 15 June 2011, I?aki Baz Castillo wrote:
> 2011/6/9 I?aki Baz Castillo <ibc at aliax.net>:
> > 2011/6/9 Alex Hermann <alex at speakup.nl>:
> > 2) About your quoted text, indeed RFC 2915 says that. But that just
> > means that, an application (in this case a SIP client) must take
> > *first* the valid NAPTR record with better "order" (lowest value). But
> > RFC 2915 is just about NAPTR, it can not mandate that applications or
> > protocols cannot try other NAPTR records as failover mechanism. Any
> > SIP client or proxy implementing NAPTR records does failover to the
> > next NAPTR record in case the first one failed (at least those I've
> > tested).
> 
> It seems that sip-router behaves exactly as you describe. This it, it
> performs the NAPTR resolution, chooses the record with best 'order'
> (for the supported transports) and just uses it (it would do SRV
> failover and so, but would never try other NAPTR records with less
> 'order').
> 
> However I've seen other SIP stacks also performing failover by taking
> next NAPTR records. But indeed, RFC 2915 does not mandate it, instead
> it says MUST NOT. So you are right.

I forgot about a discussion i had about this issue a few years ago with a 
customer and just remembered the outcome now. (Optional) failover can be 
provided by different 'Preference' field within the same 'Order'. From rfc 
2915/3403 section 2:

      The important difference between Order and Preference is that
      once a match is found the client MUST NOT consider records with a
      different Order but they MAY process records with the same Order
      but different Preferences.  I.e., Preference is used to give weight
      to rules that are considered the same from an authority
      standpoint but not from a simple load balancing standpoint.

Unfortunately this wasn't possible with Kamailio <=1.5. Have you tried 3.x 
with equal 'Order' but different 'Preference' or just with different 'Order'?

-- 
Greetings,

Alex Hermann



From omar at 321communications.com  Wed Jun 15 15:14:06 2011
From: omar at 321communications.com (Omar)
Date: Wed, 15 Jun 2011 09:14:06 -0400
Subject: [SR-Users] Operations With Variables $var- does it need to load
	a module
In-Reply-To: <4DF8A46A.6010004@gmail.com>
References: <6C5E1099-D53A-4C6D-A36C-F0DC07D20C00@321communications.com>
	<4DF8A46A.6010004@gmail.com>
Message-ID: <5CB8C0E8-8ECB-423D-83FF-43D20E7BD1C7@321communications.com>


BEFORE VAR A#################################################################===__IN FAILURE_ROUTE  :::======>  RURI=+1404345 

VAR A                #################################################################===__IN FAILURE_ROUTE  :::======>  rU = +1404345 VAR A = 8  VAR B = 0 
AFTER VAR B #################################################################===__IN FAILURE_ROUTE  :::======>  rU = +1404345 VAR A = 8  VAR B = 8 :: :: 



i changed the tU to rU, i have the same result, just to make a test.

is really interesting.

O

On Jun 15, 2011, at 8:24 AM, Daniel-Constantin Mierla wrote:

> Hello,
> 
> On 6/13/11 7:05 AM, Omar wrote:
>> In Kamailio 1.4.3
>> 
>> I have the following scenario:
>> 
>> $var(a) = 0
>> $var(b) = 0
>> $var(a) = $(tU{s.len});
>> $var(b) =  $var(a) - 5;
>> 
>> xlog(" $var(b)");
>> 
>> the result $var(b)  is always the same as $var(a)
>> 
>> should i check if some specific module is loaded?
>> the debugs does not show any error, and $var(a) and $var(b) are unique.
>> 
>> Variables are working, but not the operations with the variables.
> there is no module needed to be loaded for that version.
> 
> Can you add xlog() lines to print the values of variables for each step, including the $tU s.len transformation.
> 
> Cheers,
> Daniel
> 
> -- 
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
> 
> 



From ibc at aliax.net  Wed Jun 15 15:26:05 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Wed, 15 Jun 2011 15:26:05 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <201106151509.22242.alex@speakup.nl>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTi=ijY08owMepH4K2yDrKQrNfQZzeQ@mail.gmail.com>
	<BANLkTi=d1inRHxr2bQ+Rh4bZejg9sdFvgA@mail.gmail.com>
	<201106151509.22242.alex@speakup.nl>
Message-ID: <BANLkTikZi+X+bKdCjAwH_w=7Dzki6NNdDA@mail.gmail.com>

2011/6/15 Alex Hermann <alex at speakup.nl>:
>> However I've seen other SIP stacks also performing failover by taking
>> next NAPTR records. But indeed, RFC 2915 does not mandate it, instead
>> it says MUST NOT. So you are right.
>
> I forgot about a discussion i had about this issue a few years ago with a
> customer and just remembered the outcome now. (Optional) failover can be
> provided by different 'Preference' field within the same 'Order'. From rfc
> 2915/3403 section 2:
>
> ? ? ?The important difference between Order and Preference is that
> ? ? ?once a match is found the client MUST NOT consider records with a
> ? ? ?different Order but they MAY process records with the same Order
> ? ? ?but different Preferences. ?I.e., Preference is used to give weight
> ? ? ?to rules that are considered the same from an authority
> ? ? ?standpoint but not from a simple load balancing standpoint.

Yes, 100% right.


> Unfortunately this wasn't possible with Kamailio <=1.5. Have you tried 3.x
> with equal 'Order' but different 'Preference' or just with different 'Order'?

Not yet, but I'll do it and comment here :)

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


From mino.haluz at gmail.com  Wed Jun 15 15:42:44 2011
From: mino.haluz at gmail.com (Mino Haluz)
Date: Wed, 15 Jun 2011 15:42:44 +0200
Subject: [SR-Users] kamailio conference calls support
Message-ID: <BANLkTimAp3P6C9VjTfPvD1vz+4qpWRkcmg@mail.gmail.com>

Hi,

does kamailio support conference calls or it has to be implemented in
some B2BUA ?
If it is supported, what are the requirements (modules etc.).

Thanks,
Mino


From carsten at ng-voice.com  Wed Jun 15 15:44:05 2011
From: carsten at ng-voice.com (Carsten Bock)
Date: Wed, 15 Jun 2011 15:44:05 +0200
Subject: [SR-Users] kamailio conference calls support
In-Reply-To: <BANLkTimAp3P6C9VjTfPvD1vz+4qpWRkcmg@mail.gmail.com>
References: <BANLkTimAp3P6C9VjTfPvD1vz+4qpWRkcmg@mail.gmail.com>
Message-ID: <BANLkTim8acOjXxZn8Dg+bKcB2vVjZ4nmxA@mail.gmail.com>

Hi,

Kamailio is just a SIP-Proxy, so it has no support for mixing Audio.

Carsten

2011/6/15 Mino Haluz <mino.haluz at gmail.com>:
> Hi,
>
> does kamailio support conference calls or it has to be implemented in
> some B2BUA ?
> If it is supported, what are the requirements (modules etc.).
>
> 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
>



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

Schomburgstr. 80
22767 Hamburg
Germany

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220


From ibc at aliax.net  Wed Jun 15 16:18:41 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Wed, 15 Jun 2011 16:18:41 +0200
Subject: [SR-Users] NAPTR priorities doesn't seem to work properly
In-Reply-To: <BANLkTikZi+X+bKdCjAwH_w=7Dzki6NNdDA@mail.gmail.com>
References: <BANLkTi=ZRKusws2jxKv6AMD=B+F71gkD=w@mail.gmail.com>
	<BANLkTi=ijY08owMepH4K2yDrKQrNfQZzeQ@mail.gmail.com>
	<BANLkTi=d1inRHxr2bQ+Rh4bZejg9sdFvgA@mail.gmail.com>
	<201106151509.22242.alex@speakup.nl>
	<BANLkTikZi+X+bKdCjAwH_w=7Dzki6NNdDA@mail.gmail.com>
Message-ID: <BANLkTinijC2UFpp9kNbp4pOpORGvYviFxg@mail.gmail.com>

2011/6/15 I?aki Baz Castillo <ibc at aliax.net>:
>> Unfortunately this wasn't possible with Kamailio <=1.5. Have you tried 3.x
>> with equal 'Order' but different 'Preference' or just with different 'Order'?
>
> Not yet, but I'll do it and comment here :)

Kamailio just takes one of the NAPTR records, even if the first fails
and there are more NAPTR records with same order (tested right now).
The file doc/dns-txt confirms it:

   dns_try_naptr = on | off - if on sip-router will first try a NAPTR lookup for
      destinations that don't have the protocol or port specified and
      are not simple ip addresses (as described in RFC 3263). This will
      introduce a slight performance penalty and will probably cause extra
      DNS lookups. For example a lookup for a non-existing domain will
      produce one extra query: NAPTR(domain), SRV(_sip._udp.domain)
      and A/AAAA(domain).
      If the result of a query contains several NAPTR records,
sip-router will select
      among them according to the RFC2915 and sip-router preference towards a
      specific protocol (see dns_udp_pref, dns_tcp_pref and dns_tls_pref
      below). For an RFC3263 compliant configuration (choose the remote side
      preferred protocol if supported), set dns_udp_pref, dns_tcp_pref and
      dns_tls_pref to the same value (>=0), e.g. 0.
      Default: off

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


From realjijo at gmail.com  Wed Jun 15 17:14:00 2011
From: realjijo at gmail.com (Jijo)
Date: Wed, 15 Jun 2011 11:14:00 -0400
Subject: [SR-Users] 200 OK Retransmission Error
In-Reply-To: <4DF864AE.3010905@gmail.com>
References: <BANLkTim_TaNti=H1cJ=bGj=7a4-DWZa8=w@mail.gmail.com>
	<4DF864AE.3010905@gmail.com>
Message-ID: <BANLkTimEvsMfXO==_Fvw5Msuj=-q+JLbpQ@mail.gmail.com>

Hi Daniel,

Thanks.. Is there any side effects if we increase the wt_timer to 32secs to
match with the INVITE transaction timer?

regards,
Jijo
On Wed, Jun 15, 2011 at 3:52 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

> Hello,
>
>
> On 6/15/11 6:16 AM, Jijo wrote:
>
>
>
>  Hi All,
>
> In an INVITE transaction, no ACK from A party and B party continue
> retransmission of 200 OK. I have observed that kamailio is not passing the
> retransmitted 200 OK message from B party to the Application(script) after 3
> seconds. Instead kamailio directly sending to A party without passing to the
> script.
>
> what do you mean "without passing to the script"?
>
> Note that a transaction is completed when a final reply is forwarded back
> to the sender of request. Then it is kept fore a short time in memory, if
> you want to increase that interval, see this parameter in tm:
>
> http://kamailio.org/docs/modules/stable/modules/tm.html#wt_timer
>
> Cheers,
> Daniel
>
>
>  I see in the logs that kamailio is cleaning up the transaction after 3
> seconds..look for the string "DEBUG: t_lookup_request: no transaction
> found"  before this log it is cleaning up the timer in the log "DEBUG:
> cleanup_uac_timers: RETR/FR timers reset"
>
> What does it mean by that??
> How can we make the kamailio to pass the response to the application?
>
> Please let me know your comments.. Here is the flow. I cannot send the
> complete logs which is quite big. so i grep'd using the stirng
> "tcp_read_req:
> content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from
> the complete log..
>
> A             kamailo                 B
>
> ----INVITE-->- --------------------INVITE
>
> 180 Ringing--<---------------180 Ringing
> 200 OK-----------<---------------200 OK 0
> 200 OK-----------<---------------200 OK 500ms
> 200 OK-----------<---------------200 OK 1s
> 200 OK-----------<---------------200 OK 2s
>
> < From here it is not passing to the application- script>
>
> 200 OK-----------<---------------200 OK 4s
> 200 OK-----------<---------------200 OK 4s
>
>
>
>   28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 0
>   282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
>   292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO:
> <script>: CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null>
> From=sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=
> 5616990102 <sip:5616990102 at 192.168.243.161:5060;transport=tcp>
>   331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
>   356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO:
> <script>: CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null>
> From=sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=
> 5616990102 <sip:5616990102 at 192.168.243.161:5060;transport=tcp>
>   398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset
>   406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
>   431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO:
> <script>: CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null>
> From=sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=
> 5616990102 <sip:5616990102 at 192.168.243.161:5060;transport=tcp>
>   481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
>   506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO:
> <script>: CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null>
> From=sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=
> 5616990102 <sip:5616990102 at 192.168.243.161:5060;transport=tcp>
>   556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)!
>   581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO:
> <script>: CI=1-8028 at 192.168.243.150 -RR11 - Entry M=INVITE RURI=<null>
> From=sip:5616990103 at 192.168.243.150 To=sip:5616990102 at 192.168.243.151:5060RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=
> 5616990102 <sip:5616990102 at 192.168.243.161:5060;transport=tcp>
>   631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>   693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>   755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>   817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>   879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>   941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core>
> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>   1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [tcp_read.c:933]: tcp_read_req: content-length= 191
>   1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[
> sip:5616990102 at 192.168.243.151:5060]
>   1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG:
> <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
>   1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch
> 0
>   1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists
>   1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm
> [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From klaus.mailinglists at pernau.at  Wed Jun 15 18:19:52 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 15 Jun 2011 18:19:52 +0200
Subject: [SR-Users] Operations With Variables $var- does it need to load
 a module
In-Reply-To: <5CB8C0E8-8ECB-423D-83FF-43D20E7BD1C7@321communications.com>
References: <6C5E1099-D53A-4C6D-A36C-F0DC07D20C00@321communications.com>	<4DF8A46A.6010004@gmail.com>
	<5CB8C0E8-8ECB-423D-83FF-43D20E7BD1C7@321communications.com>
Message-ID: <4DF8DBA8.2040508@pernau.at>

I tried it in normale route and failure route, it works (at least in 
Kamailio 3.1)

regards
klaus

On 15.06.2011 15:14, Omar wrote:
>
> BEFORE VAR A#################################################################===__IN FAILURE_ROUTE  :::======>   RURI=+1404345
>
> VAR A                #################################################################===__IN FAILURE_ROUTE  :::======>   rU = +1404345 VAR A = 8  VAR B = 0
> AFTER VAR B #################################################################===__IN FAILURE_ROUTE  :::======>   rU = +1404345 VAR A = 8  VAR B = 8 :: ::
>
>
>
> i changed the tU to rU, i have the same result, just to make a test.
>
> is really interesting.
>
> O
>
> On Jun 15, 2011, at 8:24 AM, Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> On 6/13/11 7:05 AM, Omar wrote:
>>> In Kamailio 1.4.3
>>>
>>> I have the following scenario:
>>>
>>> $var(a) = 0
>>> $var(b) = 0
>>> $var(a) = $(tU{s.len});
>>> $var(b) =  $var(a) - 5;
>>>
>>> xlog(" $var(b)");
>>>
>>> the result $var(b)  is always the same as $var(a)
>>>
>>> should i check if some specific module is loaded?
>>> the debugs does not show any error, and $var(a) and $var(b) are unique.
>>>
>>> Variables are working, but not the operations with the variables.
>> there is no module needed to be loaded for that version.
>>
>> Can you add xlog() lines to print the values of variables for each step, including the $tU s.len transformation.
>>
>> 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


From omar at 321communications.com  Wed Jun 15 19:06:49 2011
From: omar at 321communications.com (Omar)
Date: Wed, 15 Jun 2011 13:06:49 -0400
Subject: [SR-Users] Operations With Variables $var- does it need to load
	a module
In-Reply-To: <4DF8DBA8.2040508@pernau.at>
References: <6C5E1099-D53A-4C6D-A36C-F0DC07D20C00@321communications.com>	<4DF8A46A.6010004@gmail.com>
	<5CB8C0E8-8ECB-423D-83FF-43D20E7BD1C7@321communications.com>
	<4DF8DBA8.2040508@pernau.at>
Message-ID: <A0F5FF65-C3B1-443F-B620-9E29521523B5@321communications.com>

sight !   :-(
thank you Klaus, We just start testing the 3.1.4 but until the Labs releases will take some time 

O

On Jun 15, 2011, at 12:19 PM, Klaus Darilion wrote:

> I tried it in normale route and failure route, it works (at least in Kamailio 3.1)
> 
> regards
> klaus
> 
> On 15.06.2011 15:14, Omar wrote:
>> 
>> BEFORE VAR A#################################################################===__IN FAILURE_ROUTE  :::======>   RURI=+1404345
>> 
>> VAR A                #################################################################===__IN FAILURE_ROUTE  :::======>   rU = +1404345 VAR A = 8  VAR B = 0
>> AFTER VAR B #################################################################===__IN FAILURE_ROUTE  :::======>   rU = +1404345 VAR A = 8  VAR B = 8 :: ::
>> 
>> 
>> 
>> i changed the tU to rU, i have the same result, just to make a test.
>> 
>> is really interesting.
>> 
>> O
>> 
>> On Jun 15, 2011, at 8:24 AM, Daniel-Constantin Mierla wrote:
>> 
>>> Hello,
>>> 
>>> On 6/13/11 7:05 AM, Omar wrote:
>>>> In Kamailio 1.4.3
>>>> 
>>>> I have the following scenario:
>>>> 
>>>> $var(a) = 0
>>>> $var(b) = 0
>>>> $var(a) = $(tU{s.len});
>>>> $var(b) =  $var(a) - 5;
>>>> 
>>>> xlog(" $var(b)");
>>>> 
>>>> the result $var(b)  is always the same as $var(a)
>>>> 
>>>> should i check if some specific module is loaded?
>>>> the debugs does not show any error, and $var(a) and $var(b) are unique.
>>>> 
>>>> Variables are working, but not the operations with the variables.
>>> there is no module needed to be loaded for that version.
>>> 
>>> Can you add xlog() lines to print the values of variables for each step, including the $tU s.len transformation.
>>> 
>>> 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
> 
> _______________________________________________
> 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 potxoka at gmail.com  Thu Jun 16 01:07:40 2011
From: potxoka at gmail.com (Anto)
Date: Thu, 16 Jun 2011 01:07:40 +0200
Subject: [SR-Users] Security in proxy
Message-ID: <4DF93B3C.9060109@gmail.com>

Hi

After reading the modules pike, pipelimit, etc. I wanted to know what 
measures can be used in the proxy, because like me, there will be more 
people interested ;-).

We see that the module pike is a good security measure, but for users 
with many channels, we used the configuration of users with few channels 
(could we discriminate and make different rules for each?). If we have 
the proxy settings for normal users and have trunk configuration as 
well, I guess we also mark the traffic of the trunk to see it too 
aggressive about the user. As you might know if discriminate different 
types of users according to their traffic, one solution I can think of 
is to have different proxies for each user type (trunk, normal, etc).

Is there any way to implement something that detects unusual behavior on 
the user (referred to countries that do not ever called, excessive 
traffic, etc)?. I ask not to try to implement something that exists or 
perhaps someone can advise me how to do so because it is very likely not 
get it or do it wrong :-P . Continue to seek information and if I find 
something to contribute, send him to the list.

Can you think of some way more secure proxy (mainly user accounts)? Do 
some basic safety tips to take into consideration? Any guidance? My 
advice to avoid trying to schedule something with 100 lines of code when 
I can do with 5 lines? Does this indicate that I intend to do not I make 
it :-P ?. What I said, I will try paper on the subject and if I can find 
information on something (though it may seem a bad code :-( ), which 
publishes. Thanks ;-)

Regards
Anto
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From hcouplet at gmail.com  Thu Jun 16 01:48:45 2011
From: hcouplet at gmail.com (Herve Couplet)
Date: Thu, 16 Jun 2011 01:48:45 +0200
Subject: [SR-Users] t_relay_to("0x02") alternative with kamailio 3.1
Message-ID: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>

Hi,

I had a setup with an old version where I used the function
t_relay_to("0x02").
Following the 3.1 documentation, flag :
*     0x02* - do not generate reply on internal error (NOTE: has no effect
anymore)

I used that function as a workaround with a pbx that stop to register when
it receive an error response like
478 Unresolvable destination or 408 request timeout.

I did upgrade to 3.1 to have dns srv caching.

Do you know how this would be possible with kamailio 3.1.

Thanks for your suggestions,
hp
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From abalashov at evaristesys.com  Thu Jun 16 01:52:47 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 15 Jun 2011 19:52:47 -0400
Subject: [SR-Users] t_relay_to("0x02") alternative with kamailio 3.1
In-Reply-To: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>
References: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>
Message-ID: <720C0A69-5E85-4F50-8C4E-1D1FAB653973@evaristesys.com>

You can catch and drop replies in an an onreply_route or failure_route.

if(t_check_status("478"))
   exit;

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jun 15, 2011, at 7:48 PM, Herve Couplet <hcouplet at gmail.com> wrote:

> Hi,
> 
> I had a setup with an old version where I used the function t_relay_to("0x02").
> Following the 3.1 documentation, flag :
>      0x02 - do not generate reply on internal error (NOTE: has no effect anymore)
> 
> I used that function as a workaround with a pbx that stop to register when it receive an error response like 
> 478 Unresolvable destination or 408 request timeout.
> 
> I did upgrade to 3.1 to have dns srv caching.
> 
> Do you know how this would be possible with kamailio 3.1.
> 
> Thanks for your suggestions,
> hp
> _______________________________________________
> 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 herve at couplet.be  Thu Jun 16 02:09:15 2011
From: herve at couplet.be (Herve Couplet)
Date: Thu, 16 Jun 2011 02:09:15 +0200
Subject: [SR-Users] t_relay_to("0x02") alternative with kamailio 3.1
In-Reply-To: <720C0A69-5E85-4F50-8C4E-1D1FAB653973@evaristesys.com>
References: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>
	<720C0A69-5E85-4F50-8C4E-1D1FAB653973@evaristesys.com>
Message-ID: <BANLkTi=DR8xoQYBsgOGtzYSCtf2-Q5yTFg@mail.gmail.com>

Already tried that.

Documentation on t_relay from 1.5 says :
*0x02* - do not internally send a negative reply in case of forward failure
(due internal error, bad RURI, bad message, etc). When a forward failure
occurs, no SIP request is relayed and therefore no negative reply or timeout
will show up on the failure_route (if one is set). It applies only when the
transaction is created. By default one negative reply is sent. Useful if you
want to implement a serial forking in case of failure.

The pbx use a dsl connection with a dynamic ip that is renewed during the
night.
So if a register renew comes at the same time there is a ip update during
the night, it will get an error 478 and will stop to register.

My setup is the following :
t_on_reply("3");
t_on_failure("9");
if (!t_relay_to("0x02"))
{
    xlog("L_INFO", "REGISTER RELAY ERROR DROP req > [$rm], from URI: [$fu],
To URI: [$tu]");
    drop();
}

onreply_route[3] {
    xlog("L_INFO", "REPLY ROUTE rep > [$rm], $rs $rr, from URI: [$fu], To
URI: [$tu]");
}

failure_route[9]
{
    xlog("L_INFO", "failure_route rep > [$rm], $rs $rr, from URI: [$fu], To
URI: [$tu]");
    if (!t_check_status("401") && !t_check_status("407"))
    {
xlog("L_INFO", "REGISTER2 RESPONSE ERROR DROP rep > [$rm], $rs $rr, from
URI: [$fu], To URI: [$tu]");
drop();
    }
}

I did put a wrong domain to force network error and I get in the log with
3.1.  (It doesn't go to the onreply_route or failure_route functions as it
was with 1.5).
Kamailio is replying with a SIP/2.0 478 Unresolvable destination (478/TM).
I got the error from t_relay_to but the drop() didn't prevent the 478 to be
sent.

Here is the log :
Jun 16 01:59:10 localhost /usr/sbin/kamailio[10304]: ERROR: <core>
[resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "
wrongdomain.com"
Jun 16 01:59:10 localhost /usr/sbin/kamailio[10304]: ERROR: tm [ut.h:318]:
failed to resolve "wrongdomain.com"
Jun 16 01:59:10 localhost /usr/sbin/kamailio[10304]: ERROR: tm
[t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches
Jun 16 01:59:10 localhost /usr/sbin/kamailio[10304]: INFO: <script>:
REGISTER RELAY ERROR DROP req > [REGISTER], from URI: [
sip:771639197 at wrongdomain.com], To URI: [sip:771639197 at wrongdomain.com]
Jun 16 01:59:36 localhost /usr/sbin/kamailio[10305]: ERROR: <core>
[resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "
wrongdomain.com"
Jun 16 01:59:36 localhost /usr/sbin/kamailio[10305]: ERROR: tm [ut.h:318]:
failed to resolve "wrongdomain.com"
Jun 16 01:59:36 localhost /usr/sbin/kamailio[10305]: ERROR: tm
[t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches
Jun 16 01:59:36 localhost /usr/sbin/kamailio[10305]: ERROR: sl
[sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination
(478/SL)

Regards,
hp

On Thu, Jun 16, 2011 at 1:52 AM, Alex Balashov <abalashov at evaristesys.com>wrote:

> You can catch and drop replies in an an onreply_route or failure_route.
>
> if(t_check_status("478"))
>   exit;
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Jun 15, 2011, at 7:48 PM, Herve Couplet <hcouplet at gmail.com> wrote:
>
> > Hi,
> >
> > I had a setup with an old version where I used the function
> t_relay_to("0x02").
> > Following the 3.1 documentation, flag :
> >      0x02 - do not generate reply on internal error (NOTE: has no effect
> anymore)
> >
> > I used that function as a workaround with a pbx that stop to register
> when it receive an error response like
> > 478 Unresolvable destination or 408 request timeout.
> >
> > I did upgrade to 3.1 to have dns srv caching.
> >
> > Do you know how this would be possible with kamailio 3.1.
> >
> > Thanks for your suggestions,
> > hp
> > _______________________________________________
> > 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 abalashov at evaristesys.com  Thu Jun 16 02:39:29 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 15 Jun 2011 20:39:29 -0400
Subject: [SR-Users] t_relay_to("0x02") alternative with kamailio 3.1
In-Reply-To: <BANLkTi=DR8xoQYBsgOGtzYSCtf2-Q5yTFg@mail.gmail.com>
References: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>	<720C0A69-5E85-4F50-8C4E-1D1FAB653973@evaristesys.com>
	<BANLkTi=DR8xoQYBsgOGtzYSCtf2-Q5yTFg@mail.gmail.com>
Message-ID: <4DF950C1.3070201@evaristesys.com>

On 06/15/2011 08:09 PM, Herve Couplet wrote:

> Kamailio is replying with a SIP/2.0 478 Unresolvable destination
> (478/TM). I got the error from t_relay_to but the drop() didn't
> prevent the 478 to be sent.

Oh, I see.  This is an internally-generated error from TM.

There is probably not much you can do here except to use stateless 
forwarding.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From alex at speakup.nl  Thu Jun 16 09:32:41 2011
From: alex at speakup.nl (Alex Hermann)
Date: Thu, 16 Jun 2011 09:32:41 +0200
Subject: [SR-Users] t_relay_to("0x02") alternative with kamailio 3.1
In-Reply-To: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>
References: <BANLkTinmFkz=3WgjhkNFVr1nHUsJyi9ajg@mail.gmail.com>
Message-ID: <201106160932.41312.alex@speakup.nl>

On Thursday 16 June 2011 01:48:45 Herve Couplet wrote:
> I had a setup with an old version where I used the function
> t_relay_to("0x02").
> Following the 3.1 documentation, flag :
> *     0x02* - do not generate reply on internal error (NOTE: has no effect
> anymore)
> 
> I used that function as a workaround with a pbx that stop to register when
> it receive an error response like
> 478 Unresolvable destination or 408 request timeout.
> 
> I did upgrade to 3.1 to have dns srv caching.
> 
> Do you know how this would be possible with kamailio 3.1.

If i read the docs correctly, in 3.1 t_relay() never sends an automatic reply, 
so you can just omit the flags.

-- 
Alex Hermann


From miconda at gmail.com  Thu Jun 16 12:06:56 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 16 Jun 2011 12:06:56 +0200
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <1308127843.1551.8.camel@localhost.localdomain>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com>
	<13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
	<4DF5E296.6070607@gmail.com>
	<1308049573.1569.6.camel@localhost.localdomain>
	<4DF8637D.3080306@gmail.com>
	<1308127843.1551.8.camel@localhost.localdomain>
Message-ID: <4DF9D5C0.8030801@gmail.com>



On 6/15/11 10:50 AM, Evgeniy Spinov wrote:
> On Wed, 2011-06-15 at 09:47 +0200, Daniel-Constantin Mierla wrote:
>> On 6/14/11 1:06 PM, Evgeniy Spinov wrote:
>>> On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:
>>>
>>> Ok, still unsolved problem.
>>>
>>> I've changed failover part of the script to use ds_next_dst(). It does
>>> what it should and signalization works fine - calls are being connected.
>>> However I've got no audio and message in logs:
>>>
>>> 7(9917) ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply
>>> from rtp proxy
>>>
>>> I've added force_rtp_proxy() after ds_next_dst() and message
>>> disappeared, however still no audio on the failover call.
>>>
>>> RTP Proxy runs on the same server and is fine for not failover calls.
>>> Success route script executes force_rtp_proxy() and fix_nated_contact()
>>> in curtain conditions and that's it.
>>>
>>> Am I missing something in my logic?
>> Try to do the rtpproxy thing in branch route -- anyhow this has nothing
>> to with dispatcher.
>>
>> Cheers,
>> Daniel
>>
> It's done on the branch route, however it seems that branch route is not
> being executed when the failure branch route is being ran.
do t_on_branch() in failure route as well.

Cheers,
Daniel

>   Anyhow, when
> the call is not passing through failure route and ds_next_dst() it is
> fine, no errors. When it goes there, error with rtp proxy in logs and no
> audio. When I add force_rtp_proxy() on success of t_relay() in failure
> route it gives no errors, but still no audio.
>
> I do not know where issue is related, and what is wrong, especially when
> this scenario worked before.
>
> Just to avoid misunderstanding, providing you a failure and a normal
> branches of the script. Failure route:
>
>          if (is_method("INVITE")
>                          &&  (isbflagset("6") || isflagset(5))) {
>                  unforce_rtp_proxy();
>          }
>
> 	... ( checks for different errors and messages output is skipped, not
> related to this issue anyway )
>
>              if( ds_next_dst() )
>              {
>                  xlog( "L_NOTICE", "[$Tf] $ci Next gateway $fU ->  $tU via
> $du\n" );
>                  if( !t_relay() )
>                  {
>                      xlog( "L_INFO", "[$Tf] $ci -- ERROR - Can not
> t_relay()\n" );
>                      return;
>                  };
>                  return;
>              }
>              else
>              {
>                       xlog( "L_INFO", "[$Tf] $ci No more routes available
> \n" );
>                  t_reply("503", "Service unavailable -- no more gateways"
> );
>                  exit;
>              };
>
>
> And success route completely:
>
>          if ((isflagset(5) || isbflagset("6"))&&  status=~"(183)|(2[0-9][0-9])") {
>                  force_rtp_proxy();
>          }
>          if (isbflagset("6")) {
>                  fix_nated_contact();
>          }
>
>
> Thank you.
>
>
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From henning.westerholt at 1und1.de  Thu Jun 16 12:44:30 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Thu, 16 Jun 2011 12:44:30 +0200
Subject: [SR-Users] Security in proxy
In-Reply-To: <4DF93B3C.9060109@gmail.com>
References: <4DF93B3C.9060109@gmail.com>
Message-ID: <201106161244.30337.henning.westerholt@1und1.de>

On Thursday 16 June 2011, Anto wrote:
>  After reading the modules pike, pipelimit, etc. I wanted to know what
> measures can be used in the proxy, because like me, there will be more
> people interested ;-).
> 
>  We see that the module pike is a good security measure, but for users with
> many channels, we used the configuration of users with few channels (could
> we discriminate and make different rules for each?). If we have the proxy
> settings for normal users and have trunk configuration as well, I guess we
> also mark the traffic of the trunk to see it too aggressive about the
> user. As you might know if discriminate different types of users according
> to their traffic, one solution I can think of is to have different proxies
> for each user type (trunk, normal, etc).

Hi Anto,

having different systems is one solution, you could also just use internally 
in one server different traffic classes for the user sets you have. 

>  Is there any way to implement something that detects unusual behavior on
> the user (referred to countries that do not ever called, excessive
> traffic, etc)?. I ask not to try to implement something that exists or
> perhaps someone can advise me how to do so because it is very likely not
> get it or do it wrong :-P . Continue to seek information and if I find
> something to contribute, send him to the list.

You should also look into the htable module for account/ password brute force 
detection. If you want to implement something w/r to the origination country 
of certain traffic, there is the geoip module which can provide you this 
informations, wich you could also combine with the modules already discussed.
 
>  Can you think of some way more secure proxy (mainly user accounts)? Do
> some basic safety tips to take into consideration? Any guidance? My advice
> to avoid trying to schedule something with 100 lines of code when I can do
> with 5 lines? Does this indicate that I intend to do not I make it :-P ?.
> What I said, I will try paper on the subject and if I can find information
> on something (though it may seem a bad code :-( ), which publishes. Thanks
> ;-)

With regards to user accounts, one thing you probably want to look at is 
password security and fraud or misuse detection logic. Smart attackers will 
find ways around your proactive measures, and you need to respond actively, 
like blocking users after you detected something malicious.

Best regards,

Henning


From ibc at aliax.net  Thu Jun 16 13:08:37 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 16 Jun 2011 13:08:37 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
Message-ID: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>

Hi, summarizing I have two pseudo-variables:

  $var(allowed_zones) = 5;
  $var(id_zone) = 2;

The binary representation of 5 (3+2), assuming 8 bits lenght, is:

  00000101

so bit 0 = 1,
     bit 1 = 0,
     bit 2 = 1
     ...

Now I just need to know if bit number 2 ($var(id_zone)) of
$var(allowed_zones) is 0 or 1.

Is there any function like isflagset() but for pseudo-variables?

Thanks a lot.



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


From ibc at aliax.net  Thu Jun 16 13:12:51 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 16 Jun 2011 13:12:51 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
In-Reply-To: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
References: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
Message-ID: <BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>

2011/6/16 I?aki Baz Castillo <ibc at aliax.net>:
> Now I just need to know if bit number 2 ($var(id_zone)) of
> $var(allowed_zones) is 0 or 1.
>
> Is there any function like isflagset() but for pseudo-variables?

I've found isavpflagset() right now:

  http://sip-router.org/wiki/cookbooks/core-cookbook/devel#isavpflagset

But it has no description. How is it supposed to work?

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


From ibc at aliax.net  Thu Jun 16 13:15:39 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 16 Jun 2011 13:15:39 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
In-Reply-To: <BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>
References: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
	<BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>
Message-ID: <BANLkTinEXiOQXkx4=XAXtNQkppqRFadARg@mail.gmail.com>

2011/6/16 I?aki Baz Castillo <ibc at aliax.net>:
> I've found isavpflagset() right now:
>
> ?http://sip-router.org/wiki/cookbooks/core-cookbook/devel#isavpflagset
>
> But it has no description. How is it supposed to work?

The Changelog file says:

-----------------------------------
          Every AVP may by flaged from script via setavpflag(avpid, flag)
          (it's similar like message flags setflags,resetflags,isflagset).
          AVP flags must be declared using avpflags statement. Flags may be
          tested via isavpflagset(avpid, flag). Both the setting and
          testing may also be processed in a module. It's currently used in
          the "rr" module for dialog_cookies. Such module should register
          flag using register_avpflag(flag_id).

          Example: avpflags myflag, dialog_cookie;

          $a = 123; setavpflag($a, "myflag");

          if (isavpflagset($a, "myflag")) { ....

          $dlg_foo = "foo"; $dlg_bar = "bar"; setavpflag("$f./^dlg_",
          "dialog_cookie");
----------------------------------------------

But is says that I must use setavpflag() to set flags in the AVP. In
my case, I retrieve the AVP value from a database (an integer field).
Could it work?


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


From miconda at gmail.com  Thu Jun 16 13:50:11 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 16 Jun 2011 13:50:11 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
In-Reply-To: <BANLkTinEXiOQXkx4=XAXtNQkppqRFadARg@mail.gmail.com>
References: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
	<BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>
	<BANLkTinEXiOQXkx4=XAXtNQkppqRFadARg@mail.gmail.com>
Message-ID: <4DF9EDF3.4090100@gmail.com>

Hello,

that function is probably for ser-like AVPs.

back to your original need, you can use directly if conditions like:

if($var(x) & 2) { ... }

will be true if the second bit is set. You have to use the power of two 
for the appropriate integer value to match the bit position. The master 
branch has bitwise left/right shift operations as well -- I was thinking 
of backporting since they were in 1.x but lost in integration of the 
core interpreter for 3.0, reported by Juha, so it is not really a new 
feature:

http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=508ad1c0011352e6d24234dbef70a3f6abbdc276

Cheers,
Daniel

On 6/16/11 1:15 PM, I?aki Baz Castillo wrote:
> 2011/6/16 I?aki Baz Castillo<ibc at aliax.net>:
>> I've found isavpflagset() right now:
>>
>>   http://sip-router.org/wiki/cookbooks/core-cookbook/devel#isavpflagset
>>
>> But it has no description. How is it supposed to work?
> The Changelog file says:
>
> -----------------------------------
>            Every AVP may by flaged from script via setavpflag(avpid, flag)
>            (it's similar like message flags setflags,resetflags,isflagset).
>            AVP flags must be declared using avpflags statement. Flags may be
>            tested via isavpflagset(avpid, flag). Both the setting and
>            testing may also be processed in a module. It's currently used in
>            the "rr" module for dialog_cookies. Such module should register
>            flag using register_avpflag(flag_id).
>
>            Example: avpflags myflag, dialog_cookie;
>
>            $a = 123; setavpflag($a, "myflag");
>
>            if (isavpflagset($a, "myflag")) { ....
>
>            $dlg_foo = "foo"; $dlg_bar = "bar"; setavpflag("$f./^dlg_",
>            "dialog_cookie");
> ----------------------------------------------
>
> But is says that I must use setavpflag() to set flags in the AVP. In
> my case, I retrieve the AVP value from a database (an integer field).
> Could it work?
>
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Thu Jun 16 14:03:52 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 16 Jun 2011 14:03:52 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
In-Reply-To: <4DF9EDF3.4090100@gmail.com>
References: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
	<BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>
	<BANLkTinEXiOQXkx4=XAXtNQkppqRFadARg@mail.gmail.com>
	<4DF9EDF3.4090100@gmail.com>
Message-ID: <BANLkTinKm0wctHt9u658UkzPbHU0jGn5Sg@mail.gmail.com>

2011/6/16 Daniel-Constantin Mierla <miconda at gmail.com>:
> that function is probably for ser-like AVPs.
>
> back to your original need, you can use directly if conditions like:
>
> if($var(x) & 2) { ... }
>
> will be true if the second bit is set. You have to use the power of two for
> the appropriate integer value to match the bit position.The master branch has bitwise left/right shift operations as well

Ok, so the above example  if($var(x) & 2) { ... } will just work
because the number 2 has the bit 2 with value 1 (well, in fact the bit
1 as the first one is the bit 0). But if I want to compare the bit 5 I
need first to get 2^5 (for the second & operator) which I can get
using bitwise left/right shift functions, am I right? (since there is
no way in kamailio to calculate 2^N directly).

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


From jgallartm at gmail.com  Thu Jun 16 14:50:55 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Thu, 16 Jun 2011 14:50:55 +0200
Subject: [SR-Users] sr-users Digest, Vol 73, Issue 40
In-Reply-To: <mailman.2824.1308060002.2672.sr-users@lists.sip-router.org>
References: <mailman.2824.1308060002.2672.sr-users@lists.sip-router.org>
Message-ID: <BANLkTimM3EnE0DTCFBcRkTagKLOLOQW5yA@mail.gmail.com>

Hello

not sure what you want to accomplish, but in case it might help this is how
we use the mtree module.
Our 3 columns are: partition, prefix, route
Partition is just a name that defines a group, this is done inside the
script before calling mt_match (via a table lookup for instance). Let's say
if a call comes from ip x we assign it to partition 400; if it comes from ip
y we assign it to partition 401. Each partition is a different tree.
Consider these entries in the tree table:
400,123456,A
401,123456,A
401,1234567,B

Let's say both x and y send a call with $rU 1234567; mt_match will return A
for ip x, and B for ip y; we use that information for taking further routing
decisions. A common situation might be: x has a unique route for a whole
country while y has that entry but has more defined routes for more specific
area codes inside that country. The column route could also be for instance
the setid of a group of gw's for the dispatcher module, etc...

Regards

Javi



Message: 3
> Date: Tue, 14 Jun 2011 14:27:53 +0200
> From: I?aki Baz Castillo <ibc at aliax.net>
> Subject: Re: [SR-Users] Use of mtree module having multiple trees in
>        the     same table
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>        Users   Mailing List" <sr-users at lists.sip-router.org>
> Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> 2011/6/13 jose luis millan <doctore96 at gmail.com>:
> > I wonder if anyone could help me understanding how can I use this
> > module in a scenario where multiple mtrees are defined behind the same
> > database table.
> >
> > I can't guess how to speficy to 'mt_match' the mtree I want to match
> > against, since the 'mtree' module parameter defines a specific memory
> > tree based on a specific table, but not based on a specific 'tname'
> > value.
>
> Indeed we have a table with these colums:
>
> - prefix:    string to identify number destination. This is the regex
> column.
> - id_zone:  integer, points to other table. This is the value column.
> - scope:    integer, can be 0 or 1.
>
> So wee need two instances of mtree:
> 1) those rows with scope 0.
> 2) those rows with scope 1.
>
> Is it possible with current mtree module? or maybe we'd do better by
> creating two views of the table (one for entries with scope 0 and
> another for entries with scope 1)?
>
> Thanks a lot.
>
> --
> I?aki Baz Castillo
> <ibc at aliax.net>
>
>
>
>
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From jgallartm at gmail.com  Thu Jun 16 14:54:00 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Thu, 16 Jun 2011 14:54:00 +0200
Subject: [SR-Users] Use of mtree module having multiple trees in the
	same table
Message-ID: <BANLkTi=6zX6zdrjXUwXK_6z=Y-xD3-Oe_g@mail.gmail.com>

Hello

not sure what you want to accomplish, but in case it might help this is how
we use the mtree module.
Our 3 columns are: partition, prefix, route
Partition is just a name that defines a group, this is done inside the
script before calling mt_match (via a table lookup for instance). Let's say
if a call comes from ip x we assign it to partition 400; if it comes from ip
y we assign it to partition 401. Each partition is a different tree.
Consider these entries in the tree table:
400,123456,A
401,123456,A
401,1234567,B

Let's say both x and y send a call with $rU 1234567; mt_match will return A
for ip x, and B for ip y; we use that information for taking further routing
decisions. A common situation might be: x has a unique route for a whole
country while y has that entry but has more defined routes for more specific
area codes inside that country. The column route could also be for instance
the setid of a group of gw's for the dispatcher module, etc...

Regards

Javi

>
> Message: 3
> > Date: Tue, 14 Jun 2011 14:27:53 +0200
> > From: I?aki Baz Castillo <ibc at aliax.net>
> > Subject: Re: [SR-Users] Use of mtree module having multiple trees in
> >        the     same table
> > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> >        Users   Mailing List" <sr-users at lists.sip-router.org>
> > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww at mail.gmail.com>
> > Content-Type: text/plain; charset=UTF-8
> >
> > 2011/6/13 jose luis millan <doctore96 at gmail.com>:
> > > I wonder if anyone could help me understanding how can I use this
> > > module in a scenario where multiple mtrees are defined behind the same
> > > database table.
> > >
> > > I can't guess how to speficy to 'mt_match' the mtree I want to match
> > > against, since the 'mtree' module parameter defines a specific memory
> > > tree based on a specific table, but not based on a specific 'tname'
> > > value.
> >
> > Indeed we have a table with these colums:
> >
> > - prefix:    string to identify number destination. This is the regex
> > column.
> > - id_zone:  integer, points to other table. This is the value column.
> > - scope:    integer, can be 0 or 1.
> >
> > So wee need two instances of mtree:
> > 1) those rows with scope 0.
> > 2) those rows with scope 1.
> >
> > Is it possible with current mtree module? or maybe we'd do better by
> > creating two views of the table (one for entries with scope 0 and
> > another for entries with scope 1)?
> >
> > Thanks a lot.
> >
> > --
> > I?aki Baz Castillo
> > <ibc at aliax.net>
> >
> >
> >
> >
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> ------------------------------
>
> _______________________________________________
> sr-users mailing list
> sr-users at lists.sip-router.org
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From ibc at aliax.net  Thu Jun 16 15:29:05 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 16 Jun 2011 15:29:05 +0200
Subject: [SR-Users] Use of mtree module having multiple trees in the
	same table
In-Reply-To: <BANLkTi=6zX6zdrjXUwXK_6z=Y-xD3-Oe_g@mail.gmail.com>
References: <BANLkTi=6zX6zdrjXUwXK_6z=Y-xD3-Oe_g@mail.gmail.com>
Message-ID: <BANLkTin67hTLwM10h06Po6z_gVknJSUe5A@mail.gmail.com>

2011/6/16 Javier Gallart <jgallartm at gmail.com>:
> not sure what you want to accomplish, but in case it might help this is how
> we use the mtree module.
> Our 3 columns are: partition, prefix, route
> Partition is just a name that defines a group, this is done inside the
> script before calling mt_match (via a table lookup for instance). Let's say
> if a call comes from ip x we assign it to partition 400; if it comes from ip
> y we assign it to partition 401. Each partition is a different tree.
> Consider these entries in the tree table:
> 400,123456,A
> 401,123456,A
> 401,1234567,B
>
> Let's say both x and y send a call with $rU 1234567; mt_match will return A
> for ip x, and B for ip y; we use that information for taking further routing
> decisions. A common situation might be: x has a unique route for a whole
> country while y has that entry but has more defined routes for more specific
> area codes inside that country. The column route could also be for instance
> the setid of a group of gw's for the dispatcher module, etc...


Hi Javier, indeed this is very similar to the usage we give now to
mtree, by using different "partitions" of the table. We have it
already working (as Jose Luis told in his last mail in this thread) :)

Thanks a lot.

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


From miconda at gmail.com  Thu Jun 16 16:15:06 2011
From: miconda at gmail.com (Daniel-Constantine Mierla)
Date: Thu, 16 Jun 2011 16:15:06 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
In-Reply-To: <BANLkTinKm0wctHt9u658UkzPbHU0jGn5Sg@mail.gmail.com>
References: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
	<BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>
	<BANLkTinEXiOQXkx4=XAXtNQkppqRFadARg@mail.gmail.com>
	<4DF9EDF3.4090100@gmail.com>
	<BANLkTinKm0wctHt9u658UkzPbHU0jGn5Sg@mail.gmail.com>
Message-ID: <6B88E774-9E1A-4CD7-A9FA-11216AE989A0@gmail.com>



On Jun 16, 2011, at 2:03 PM, I?aki Baz Castillo <ibc at aliax.net> wrote:

> 2011/6/16 Daniel-Constantin Mierla <miconda at gmail.com>:
>> that function is probably for ser-like AVPs.
>> 
>> back to your original need, you can use directly if conditions like:
>> 
>> if($var(x) & 2) { ... }
>> 
>> will be true if the second bit is set. You have to use the power of two for
>> the appropriate integer value to match the bit position.The master branch has bitwise left/right shift operations as well
> 
> Ok, so the above example  if($var(x) & 2) { ... } will just work
> because the number 2 has the bit 2 with value 1 (well, in fact the bit
> 1 as the first one is the bit 0). But if I want to compare the bit 5 I
> need first to get 2^5 (for the second & operator) which I can get
> using bitwise left/right shift functions, am I right? (since there is
> no way in kamailio to calculate 2^N directly).

You can do 2*2*2... to get the power of two. Doing it in a while loop gives dynamic power parameter. Other option is to use Lua for example.

... until the left shift gets in 3.1 - if you can test the patch from master and all goes fine, then it may happen faster :-)

Cheers,
Daniel
> 


From ibc at aliax.net  Thu Jun 16 16:29:24 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 16 Jun 2011 16:29:24 +0200
Subject: [SR-Users] How to check a bit (1/0) of a pseudo-variable?
In-Reply-To: <6B88E774-9E1A-4CD7-A9FA-11216AE989A0@gmail.com>
References: <BANLkTinXcpaxb3uGJxNtfFNF8KN8k1CO1Q@mail.gmail.com>
	<BANLkTinr12vxC+9j=z0dK1FjVdDAZvimbg@mail.gmail.com>
	<BANLkTinEXiOQXkx4=XAXtNQkppqRFadARg@mail.gmail.com>
	<4DF9EDF3.4090100@gmail.com>
	<BANLkTinKm0wctHt9u658UkzPbHU0jGn5Sg@mail.gmail.com>
	<6B88E774-9E1A-4CD7-A9FA-11216AE989A0@gmail.com>
Message-ID: <BANLkTik5bFCLDVCo5bACDHJU+uXOGmQjHg@mail.gmail.com>

2011/6/16 Daniel-Constantine Mierla <miconda at gmail.com>:
> You can do 2*2*2... to get the power of two. Doing it in a while loop gives dynamic power parameter. Other option is to use Lua for example.

A bit exotic just to calculate a simple mathematical operation :)


> ... until the left shift gets in 3.1 - if you can test the patch from master and all goes fine, then it may happen faster :-)

We have it already working using that ;)

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


From lucasaa at gmail.com  Thu Jun 16 16:33:44 2011
From: lucasaa at gmail.com (Lucas Alvarez)
Date: Thu, 16 Jun 2011 11:33:44 -0300
Subject: [SR-Users] Invite replies
Message-ID: <BANLkTinocH0337wuSxCWTVyAY5a52NT+MQ@mail.gmail.com>

Hi, how may I catch the reply of an invite? I did this but it seems is not
working:

 if (method=="INVITE") {
                                $avp(sdp_ret) = fix_nated_sdp("11");
                                xlog("L_ALERT","#####********###### SDP
return: $avp(sdp_ret)\n");
                                append_hf("P-hint: sdp rewritten\r\n");
                                t_on_reply("REPLY_TWO");
}

onreply_route("REPLY_TWO") {
                                xlog("L_ALERT","ALERT: ################  $rU
- $rm from $fu (IP:$si:$sp--------$di)\n");

                                return;
}


Any help will be appreciate it, thanks in advance.

Lucas
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From spinov_evgeniy at intalisan.com  Thu Jun 16 17:12:35 2011
From: spinov_evgeniy at intalisan.com (Evgeniy Spinov)
Date: Thu, 16 Jun 2011 16:12:35 +0100
Subject: [SR-Users] Kamailio dispatcher
In-Reply-To: <4DF9D5C0.8030801@gmail.com>
References: <1307724203.1565.11.camel@localhost.localdomain>
	<4DF25D89.9090104@gmail.com> <13993171.20110612235614@intalisan.com>
	<4DF5CF19.7090205@gmail.com>
	<1307959404.2074.26.camel@localhost.localdomain>
	<4DF5E296.6070607@gmail.com>
	<1308049573.1569.6.camel@localhost.localdomain>
	<4DF8637D.3080306@gmail.com>
	<1308127843.1551.8.camel@localhost.localdomain>
	<4DF9D5C0.8030801@gmail.com>
Message-ID: <1308237155.3170.1.camel@localhost.localdomain>

On Thu, 2011-06-16 at 12:06 +0200, Daniel-Constantin Mierla wrote:
> 
> On 6/15/11 10:50 AM, Evgeniy Spinov wrote:
> > On Wed, 2011-06-15 at 09:47 +0200, Daniel-Constantin Mierla wrote:
> >> On 6/14/11 1:06 PM, Evgeniy Spinov wrote:
> >>> On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:
> >>>
> >>> Ok, still unsolved problem.
> >>>
> >>> I've changed failover part of the script to use ds_next_dst(). It does
> >>> what it should and signalization works fine - calls are being connected.
> >>> However I've got no audio and message in logs:
> >>>
> >>> 7(9917) ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply
> >>> from rtp proxy
> >>>
> >>> I've added force_rtp_proxy() after ds_next_dst() and message
> >>> disappeared, however still no audio on the failover call.
> >>>
> >>> RTP Proxy runs on the same server and is fine for not failover calls.
> >>> Success route script executes force_rtp_proxy() and fix_nated_contact()
> >>> in curtain conditions and that's it.
> >>>
> >>> Am I missing something in my logic?
> >> Try to do the rtpproxy thing in branch route -- anyhow this has nothing
> >> to with dispatcher.
> >>
> >> Cheers,
> >> Daniel
> >>
> > It's done on the branch route, however it seems that branch route is not
> > being executed when the failure branch route is being ran.
> do t_on_branch() in failure route as well.


Tried that - nothing changes. Getting "incorrect port 0 in reply from
rtpproxy" and no audio.




> 
> Cheers,
> Daniel
> 
> >   Anyhow, when
> > the call is not passing through failure route and ds_next_dst() it is
> > fine, no errors. When it goes there, error with rtp proxy in logs and no
> > audio. When I add force_rtp_proxy() on success of t_relay() in failure
> > route it gives no errors, but still no audio.
> >
> > I do not know where issue is related, and what is wrong, especially when
> > this scenario worked before.
> >
> > Just to avoid misunderstanding, providing you a failure and a normal
> > branches of the script. Failure route:
> >
> >          if (is_method("INVITE")
> >                          &&  (isbflagset("6") || isflagset(5))) {
> >                  unforce_rtp_proxy();
> >          }
> >
> > 	... ( checks for different errors and messages output is skipped, not
> > related to this issue anyway )
> >
> >              if( ds_next_dst() )
> >              {
> >                  xlog( "L_NOTICE", "[$Tf] $ci Next gateway $fU ->  $tU via
> > $du\n" );
> >                  if( !t_relay() )
> >                  {
> >                      xlog( "L_INFO", "[$Tf] $ci -- ERROR - Can not
> > t_relay()\n" );
> >                      return;
> >                  };
> >                  return;
> >              }
> >              else
> >              {
> >                       xlog( "L_INFO", "[$Tf] $ci No more routes available
> > \n" );
> >                  t_reply("503", "Service unavailable -- no more gateways"
> > );
> >                  exit;
> >              };
> >
> >
> > And success route completely:
> >
> >          if ((isflagset(5) || isbflagset("6"))&&  status=~"(183)|(2[0-9][0-9])") {
> >                  force_rtp_proxy();
> >          }
> >          if (isbflagset("6")) {
> >                  fix_nated_contact();
> >          }
> >
> >
> > Thank you.
> >
> >
> >
> 




From 00asgaroth00 at gmail.com  Thu Jun 16 18:46:02 2011
From: 00asgaroth00 at gmail.com (Asgaroth)
Date: Thu, 16 Jun 2011 17:46:02 +0100
Subject: [SR-Users] mi_datagram usage example
Message-ID: <4DFA334A.9040701@gmail.com>

Hi All,

I am trying to send a mi command over the mi_datagram udp socket. I dont
seem to be getting a response back from the server. I am certain that it
is something I am doing wrong.

I have mi_datagram loaded and listening on udp:1.1.1.1:8080.

I then try to send the following dispatcher mi command over to the mi_dg
interface:

echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080

I dont see any error in the log file, so it appears that the mi_datagram
is processing the request, but I am not getting any response back.

The ds_list command works properly if I run it using "kamctl fifo
ds_list", or if I go via "sercmd mi_dg ds_list" that works too.

Would anyone have an example bash/ksh/perl snippet that would help me on
my way to communicating with this mi interface.

Any thoughts comments would be greatly appreciated.

Thanks


From osas at voipembedded.com  Thu Jun 16 19:02:01 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Thu, 16 Jun 2011 13:02:01 -0400
Subject: [SR-Users] mi_datagram usage example
In-Reply-To: <4DFA334A.9040701@gmail.com>
References: <4DFA334A.9040701@gmail.com>
Message-ID: <BANLkTi=pLv49RnsznMoO5=HZty6yneVgKw@mail.gmail.com>

Perform a tcpdump of the traffic between nc and kamailio.
IIRC, the server replies but from a different port and therefor the
reply is discarded by nc (at least this was the issue with openser 1.x
and I assume that the same issue is present with sip-router).

See the old reports for openser here:
https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2158069&group_id=139143
https://sourceforge.net/tracker/?func=detail&atid=1086413&aid=2166279&group_id=232389


Regards,
Ovidiu Sas

On Thu, Jun 16, 2011 at 12:46 PM, Asgaroth <00asgaroth00 at gmail.com> wrote:
> Hi All,
>
> I am trying to send a mi command over the mi_datagram udp socket. I dont
> seem to be getting a response back from the server. I am certain that it
> is something I am doing wrong.
>
> I have mi_datagram loaded and listening on udp:1.1.1.1:8080.
>
> I then try to send the following dispatcher mi command over to the mi_dg
> interface:
>
> echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080
>
> I dont see any error in the log file, so it appears that the mi_datagram
> is processing the request, but I am not getting any response back.
>
> The ds_list command works properly if I run it using "kamctl fifo
> ds_list", or if I go via "sercmd mi_dg ds_list" that works too.
>
> Would anyone have an example bash/ksh/perl snippet that would help me on
> my way to communicating with this mi interface.
>
> Any thoughts comments would be greatly appreciated.
>
> 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
>


From bruce.mcalister at blueface.ie  Thu Jun 16 19:18:25 2011
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Thu, 16 Jun 2011 18:18:25 +0100
Subject: [SR-Users] mi_datagram usage example
In-Reply-To: <BANLkTi=pLv49RnsznMoO5=HZty6yneVgKw@mail.gmail.com>
References: <4DFA334A.9040701@gmail.com>
	<BANLkTi=pLv49RnsznMoO5=HZty6yneVgKw@mail.gmail.com>
Message-ID: <4DFA3AE1.1040003@blueface.ie>

Ahh yes, you are correct, it appears to be responding from a different port:

[root at myhost sbin]# ngrep -q -t -P "" -W byline -d any host 1.1.1.1 and udp
interface: any
filter: (ip) and ( host 1.1.1.1 and udp )

U 2011/06/16 18:07:56.206350 1.1.1.2:59267 -> 1.1.1.1:8080
:ds_list:



U 2011/06/16 18:07:56.208460 1.1.1.1:51384 -> 1.1.1.2:59267
200 OK
SET_NO:: 1
SET:: 1
URI:: sip:2.2.2.2:6060 flag=A priority=0 attrs=

I see in the comments from links you provided that symetric udp exchange
was a feature request, do you know if this was ever requested as a new
feature?

Thanks


On 16/06/2011 18:02, Ovidiu Sas wrote:
> Perform a tcpdump of the traffic between nc and kamailio.
> IIRC, the server replies but from a different port and therefor the
> reply is discarded by nc (at least this was the issue with openser 1.x
> and I assume that the same issue is present with sip-router).
>
> See the old reports for openser here:
> https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2158069&group_id=139143
> https://sourceforge.net/tracker/?func=detail&atid=1086413&aid=2166279&group_id=232389
>
>
> Regards,
> Ovidiu Sas
>
> On Thu, Jun 16, 2011 at 12:46 PM, Asgaroth <00asgaroth00 at gmail.com> wrote:
>> Hi All,
>>
>> I am trying to send a mi command over the mi_datagram udp socket. I dont
>> seem to be getting a response back from the server. I am certain that it
>> is something I am doing wrong.
>>
>> I have mi_datagram loaded and listening on udp:1.1.1.1:8080.
>>
>> I then try to send the following dispatcher mi command over to the mi_dg
>> interface:
>>
>> echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080
>>
>> I dont see any error in the log file, so it appears that the mi_datagram
>> is processing the request, but I am not getting any response back.
>>
>> The ds_list command works properly if I run it using "kamctl fifo
>> ds_list", or if I go via "sercmd mi_dg ds_list" that works too.
>>
>> Would anyone have an example bash/ksh/perl snippet that would help me on
>> my way to communicating with this mi interface.
>>
>> Any thoughts comments would be greatly appreciated.
>>
>> 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
>>
> _______________________________________________
> 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 mgwilliams at gmail.com  Thu Jun 16 19:51:21 2011
From: mgwilliams at gmail.com (Matthew Williams)
Date: Thu, 16 Jun 2011 13:51:21 -0400
Subject: [SR-Users] HTTP POST
Message-ID: <BANLkTin=pcQjtNyysS8P7SrtdFs=cYtn7g@mail.gmail.com>

Hi,

I'm working on an application that needs to create an HTTP POST from within
Kamailio. The utils module has the http_query function, but that is only for
GET. I think I'm looking for something more like the uac module's
uac_req_send -- but for http. Any suggestions?

Thanks in advance!
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From osas at voipembedded.com  Thu Jun 16 21:03:56 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Thu, 16 Jun 2011 15:03:56 -0400
Subject: [SR-Users] mi_datagram usage example
In-Reply-To: <4DFA3AE1.1040003@blueface.ie>
References: <4DFA334A.9040701@gmail.com>
	<BANLkTi=pLv49RnsznMoO5=HZty6yneVgKw@mail.gmail.com>
	<4DFA3AE1.1040003@blueface.ie>
Message-ID: <BANLkTi=LZs9SOh6n=cHtFfvzKfTRh9reeg@mail.gmail.com>

IIRC, the feature was requested during openser project (before the
merge with sip-router) and it was lost after the merge.
You can open a new one on the sip-router tracker.

Regards,
Ovidiu Sas

On Thu, Jun 16, 2011 at 1:18 PM, Bruce McAlister
<bruce.mcalister at blueface.ie> wrote:
> Ahh yes, you are correct, it appears to be responding from a different port:
>
> [root at myhost sbin]# ngrep -q -t -P "" -W byline -d any host 1.1.1.1 and udp
> interface: any
> filter: (ip) and ( host 1.1.1.1 and udp )
>
> U 2011/06/16 18:07:56.206350 1.1.1.2:59267 -> 1.1.1.1:8080
> :ds_list:
>
>
>
> U 2011/06/16 18:07:56.208460 1.1.1.1:51384 -> 1.1.1.2:59267
> 200 OK
> SET_NO:: 1
> SET:: 1
> URI:: sip:2.2.2.2:6060 flag=A priority=0 attrs=
>
> I see in the comments from links you provided that symetric udp exchange
> was a feature request, do you know if this was ever requested as a new
> feature?
>
> Thanks
>
>
> On 16/06/2011 18:02, Ovidiu Sas wrote:
>> Perform a tcpdump of the traffic between nc and kamailio.
>> IIRC, the server replies but from a different port and therefor the
>> reply is discarded by nc (at least this was the issue with openser 1.x
>> and I assume that the same issue is present with sip-router).
>>
>> See the old reports for openser here:
>> https://sourceforge.net/tracker/?func=detail&atid=743020&aid=2158069&group_id=139143
>> https://sourceforge.net/tracker/?func=detail&atid=1086413&aid=2166279&group_id=232389
>>
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Thu, Jun 16, 2011 at 12:46 PM, Asgaroth <00asgaroth00 at gmail.com> wrote:
>>> Hi All,
>>>
>>> I am trying to send a mi command over the mi_datagram udp socket. I dont
>>> seem to be getting a response back from the server. I am certain that it
>>> is something I am doing wrong.
>>>
>>> I have mi_datagram loaded and listening on udp:1.1.1.1:8080.
>>>
>>> I then try to send the following dispatcher mi command over to the mi_dg
>>> interface:
>>>
>>> echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080
>>>
>>> I dont see any error in the log file, so it appears that the mi_datagram
>>> is processing the request, but I am not getting any response back.
>>>
>>> The ds_list command works properly if I run it using "kamctl fifo
>>> ds_list", or if I go via "sercmd mi_dg ds_list" that works too.
>>>
>>> Would anyone have an example bash/ksh/perl snippet that would help me on
>>> my way to communicating with this mi interface.
>>>
>>> Any thoughts comments would be greatly appreciated.
>>>
>>> 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
>>>
>> _______________________________________________
>> 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 4lists at gmail.com  Thu Jun 16 22:21:55 2011
From: 4lists at gmail.com (Edson - Lists)
Date: Thu, 16 Jun 2011 17:21:55 -0300
Subject: [SR-Users] HTTP POST
In-Reply-To: <BANLkTin=pcQjtNyysS8P7SrtdFs=cYtn7g@mail.gmail.com>
References: <BANLkTin=pcQjtNyysS8P7SrtdFs=cYtn7g@mail.gmail.com>
Message-ID: <4DFA65E3.9040403@gmail.com>

Lua.

Em 16/06/2011 14:51, Matthew Williams escreveu:
> Hi,
>
> I'm working on an application that needs to create an HTTP POST from
> within Kamailio. The utils module has the http_query function, but that
> is only for GET. I think I'm looking for something more like the uac
> module's uac_req_send -- but for http. Any suggestions?
>
> 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 s.felici at mclink.eu  Fri Jun 17 11:50:15 2011
From: s.felici at mclink.eu (Simone Felici)
Date: Fri, 17 Jun 2011 11:50:15 +0200
Subject: [SR-Users] Help find out where to modify request uri
Message-ID: <4DFB2357.60802@mclink.eu>


Hi to all,

I need to edit our ser installation (OLD ser-2.0.0-rc1) implementing a check to rewrite the INVITE 
URI address in case a flag on a DB is set or not. I'm not proof on this and as always the solution 
is to find out on the production system without the possibility (for now) to upgrade to most recent 
version. In fact I would load the flag with the db_ops module.
The scenario should be the follow:
1. a call is coming from sip or from pstn to an internal sip account
2. the ser should check if a given flag for the called account is set (db_ops)
3. if it's set it should take the destination URI (taken from location db table) and rewrite the 
number to another known number. i.e. from sip:123456789 at IP;uniqu=... to sip:new987654321 at IP;uniq=...
My first issue is to identify WHERE ser execute the INVITE to the sip account and how rewrite the URI.

Could someone point me to the right place and give me some suggestion?

The ser.cfg used could be read from a previous post of me:

http://www.mail-archive.com/sr-users at lists.sip-router.org/msg04900.html


Thank's a lot!

Simon


From 4lists at gmail.com  Fri Jun 17 15:40:34 2011
From: 4lists at gmail.com (Edson - Lists)
Date: Fri, 17 Jun 2011 10:40:34 -0300
Subject: [SR-Users] Help find out where to modify request uri
In-Reply-To: <4DFB2357.60802@mclink.eu>
References: <4DFB2357.60802@mclink.eu>
Message-ID: <4DFB5952.7090705@gmail.com>

Simone,

At a first glance, make Your magic on the route[INBOUD], just after 
ALIAS lookup.

Didn't see further, but you should make the same some point before use 
of some forward function (t_relay,p.ex.)

Edson.

P.S.: Your configuration file is a mess to read, since you are asking us 
to read, understand and provide solutions to _your_ problems, could you, 
at least, provide some indentation and clean all unused and commented 
command lines??? It would help....

Em 17/06/2011 06:50, Simone Felici escreveu:
>
> Hi to all,
>
> I need to edit our ser installation (OLD ser-2.0.0-rc1) implementing a
> check to rewrite the INVITE URI address in case a flag on a DB is set or
> not. I'm not proof on this and as always the solution is to find out on
> the production system without the possibility (for now) to upgrade to
> most recent version. In fact I would load the flag with the db_ops module.
> The scenario should be the follow:
> 1. a call is coming from sip or from pstn to an internal sip account
> 2. the ser should check if a given flag for the called account is set
> (db_ops)
> 3. if it's set it should take the destination URI (taken from location
> db table) and rewrite the number to another known number. i.e. from
> sip:123456789 at IP;uniqu=... to sip:new987654321 at IP;uniq=...
> My first issue is to identify WHERE ser execute the INVITE to the sip
> account and how rewrite the URI.
>
> Could someone point me to the right place and give me some suggestion?
>
> The ser.cfg used could be read from a previous post of me:
>
> http://www.mail-archive.com/sr-users at lists.sip-router.org/msg04900.html
>
>
> Thank's a lot!
>
> 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
>


From dotnetdub at gmail.com  Fri Jun 17 16:41:51 2011
From: dotnetdub at gmail.com (dotnetdub)
Date: Fri, 17 Jun 2011 15:41:51 +0100
Subject: [SR-Users] Remote Party ID Header
Message-ID: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>

Hi All,

If an endpoint is sending us the remote party ID header, I know how to
extract $(re{uri.user}) - how would I check if the Privacy Flag was true in
Kamailio script?

Version 1.41

Thanks,
Brian
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From s.felici at mclink.eu  Fri Jun 17 16:49:25 2011
From: s.felici at mclink.eu (Simone Felici)
Date: Fri, 17 Jun 2011 16:49:25 +0200
Subject: [SR-Users] Help find out where to modify request uri
In-Reply-To: <4DFB5952.7090705@gmail.com>
References: <4DFB2357.60802@mclink.eu> <4DFB5952.7090705@gmail.com>
Message-ID: <4DFB6975.3000404@mclink.eu>


Thank you for your answer.
You've absolutely right on the linked ser.cfg. I send a cleaned up and indented version.
It's attached.
Meanwhile I'll try out a little your solution.

Thank's

Simon


Il 17/06/2011 15:40, Edson - Lists ha scritto:
> Simone,
>
> At a first glance, make Your magic on the route[INBOUD], just after ALIAS lookup.
>
> Didn't see further, but you should make the same some point before use of some forward function
> (t_relay,p.ex.)
>
> Edson.
>
> P.S.: Your configuration file is a mess to read, since you are asking us to read, understand and
> provide solutions to _your_ problems, could you, at least, provide some indentation and clean all
> unused and commented command lines??? It would help....
>
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From abalashov at evaristesys.com  Fri Jun 17 19:22:23 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 17 Jun 2011 13:22:23 -0400
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
Message-ID: <4DFB8D4F.5070506@evaristesys.com>

Brian,

Remote-Party-ID is a long-obsolete draft that never became an RFC; 
why are you still using it?

Anyway, if you know the 'privacy' header parameter will always be 
there, you can do this, assuming is_present_hf("Remote-Party-ID"), of 
course.

    $var(x) = $(hdr(Remote-Party-ID){s.select,1,>});

    xlog("L_INFO", "Remote Party ID privacy value: 
$(var(x){param.value,privacy})\n");

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From sos at sokhapkin.dyndns.org  Fri Jun 17 19:32:04 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 17 Jun 2011 13:32:04 -0400
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <4DFB8D4F.5070506@evaristesys.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
	<4DFB8D4F.5070506@evaristesys.com>
Message-ID: <201106171332.04877.sos@sokhapkin.dyndns.org>

RPID header is sent by linksys/sipura ATAs... I have a lot of headaches with 
those wide spread ATAs here in USA... PAP series are OK, but SPA series send 
RPID.

On Friday 17 June 2011, Alex Balashov wrote:
> Brian,
> 
> Remote-Party-ID is a long-obsolete draft that never became an RFC;
> why are you still using it?
> 
> Anyway, if you know the 'privacy' header parameter will always be
> there, you can do this, assuming is_present_hf("Remote-Party-ID"), of
> course.
> 
>     $var(x) = $(hdr(Remote-Party-ID){s.select,1,>});
> 
>     xlog("L_INFO", "Remote Party ID privacy value:
> $(var(x){param.value,privacy})\n");



From dotnetdub at gmail.com  Fri Jun 17 19:52:22 2011
From: dotnetdub at gmail.com (dotnetdub)
Date: Fri, 17 Jun 2011 18:52:22 +0100
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <4DFB8D4F.5070506@evaristesys.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
	<4DFB8D4F.5070506@evaristesys.com>
Message-ID: <BANLkTinFor44VSid+0aKnGeWEvM-O2R6Mw@mail.gmail.com>

On 17 June 2011 18:22, Alex Balashov <abalashov at evaristesys.com> wrote:

> Brian,
>
> Remote-Party-ID is a long-obsolete draft that never became an RFC; why are
> you still using it?
>
>

Carriers in Europe seem to still like to use it..



> Anyway, if you know the 'privacy' header parameter will always be there,
> you can do this, assuming is_present_hf("Remote-Party-**ID"), of course.
>
>   $var(x) = $(hdr(Remote-Party-ID){s.**select,1,>});
>
>   xlog("L_INFO", "Remote Party ID privacy value:
> $(var(x){param.value,privacy})**\n");
>
>
Great, thanks, will try later.

Brian



> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.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 omar at 321communications.com  Sat Jun 18 03:37:21 2011
From: omar at 321communications.com (Omar)
Date: Fri, 17 Jun 2011 21:37:21 -0400
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
Message-ID: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>



I am Using Solaris 10 and GCC 4.2, i do have libconfuse

everything goes fine until this error comes up.



carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a function)
carrierroute.c:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cmds'
carrierroute.c:132: error: variable 'exports' has initializer but incomplete type
carrierroute.c:133: warning: excess elements in struct initializer
carrierroute.c:133: warning: (near initialization for 'exports')
carrierroute.c:134: warning: excess elements in struct initializer
carrierroute.c:134: warning: (near initialization for 'exports')
carrierroute.c:135: error: 'cmds' undeclared here (not in a function)
carrierroute.c:135: warning: excess elements in struct initializer
carrierroute.c:135: warning: (near initialization for 'exports')
carrierroute.c:136: warning: excess elements in struct initializer
carrierroute.c:136: warning: (near initialization for 'exports')
carrierroute.c:137: warning: excess elements in struct initializer
carrierroute.c:137: warning: (near initialization for 'exports')
carrierroute.c:138: warning: excess elements in struct initializer
carrierroute.c:138: warning: (near initialization for 'exports')
carrierroute.c:139: warning: excess elements in struct initializer
carrierroute.c:139: warning: (near initialization for 'exports')
carrierroute.c:140: warning: excess elements in struct initializer
carrierroute.c:140: warning: (near initialization for 'exports')
carrierroute.c:141: warning: excess elements in struct initializer
carrierroute.c:141: warning: (near initialization for 'exports')
carrierroute.c:142: warning: excess elements in struct initializer
carrierroute.c:142: warning: (near initialization for 'exports')
carrierroute.c:143: warning: excess elements in struct initializer
carrierroute.c:143: warning: (near initialization for 'exports')
carrierroute.c:145: warning: excess elements in struct initializer
carrierroute.c:145: warning: (near initialization for 'exports')
carrierroute.c: In function 'mod_init':
carrierroute.c:161: error: invalid use of undefined type 'struct module_exports'
make[1]: *** [carrierroute.o] Error 1
make: *** [modules] Error 1





From a.afzali2003 at gmail.com  Sat Jun 18 15:25:04 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 18 Jun 2011 17:55:04 +0430
Subject: [SR-Users] an IDEA
Message-ID: <BANLkTiktPJ4Oyzb6isep8ZSL03-Co3BHaw@mail.gmail.com>

Hi Guys,

I need to integrate a SIP processing environment to H.248 and SS7/ ISUP
links.
For those communication links, OpenSS7 and Erlang/OTP MEGACO are good
solutions. My idea is to write applications over OTP and SIP-Router to be
able to bridge non-SIP protocols to SIP!
On OTP the way is clear, but on SIP-Router I need to be able to generate /
terminate SIP transactions. I'll appreciate every comment.

BEST,
-- afshin
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From abalashov at evaristesys.com  Sat Jun 18 23:05:02 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sat, 18 Jun 2011 17:05:02 -0400
Subject: [SR-Users] 100rel without direct reachability
Message-ID: <4DFD12FE.2090605@evaristesys.com>

(Sorry for cross-posting from Sip-Implementors;  however, upon some 
reflection this is really more of a Kamailio mechanics question.)

Greetings,

I have a scenario where endpoint A is calling endpoint B through proxy
P.  Endpoints A and B are on different networks that are not directly
reachable--A is on the public Internet, to which proxy P is connected,
but NAT'd, and B is on a private RFC1918 network on which the proxy P
has an interface as well.

The purpose of proxy P is to make calling A <-> B possible, while
providing the appropriate far-end NAT traversal functionality for A
and so on and so on.  It does this by inserting the correct
Record-Route headers into the INVITE from A, and otherwise by staying
in the signaling path;  both A and B can reach the proxy, so all
requests and replies go through it.

The problem I'm having is prior to full dialog establishment.  Upon
receipt of INVITE from A by way of proxy P, B replies with 183 + SDP
and also:

    Require: 100rel

In the 183 it provides a Contact on its private network that A
obviously cannot reach:

    Contact: <sip:xxxxx at 192.168.0.10:5060>

As a result, A attempts to send PRACKs to sip:xxxxx at 192.168.0.10,
which are obviously not going to go anywhere.  After not receiving a
few of these expected PRACKs, B replies to the call with a final
negative failure.

The question is, is there anything I can twiddle on the proxy to make
this work?  The problem with PRACKs is that although they are
sequential requests, they arise (in this case) prior to the point at
which RR can be used to guide them through the proxy.

The most obvious answer is, "Make B not Require: 100rel", of course.
However, that is not an option here.

Thanks!

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From potxoka at gmail.com  Sat Jun 18 23:05:55 2011
From: potxoka at gmail.com (Anto)
Date: Sat, 18 Jun 2011 23:05:55 +0200
Subject: [SR-Users] Security in proxy
In-Reply-To: <201106161244.30337.henning.westerholt@1und1.de>
References: <4DF93B3C.9060109@gmail.com>
	<201106161244.30337.henning.westerholt@1und1.de>
Message-ID: <4DFD1333.6070908@gmail.com>

El 16/06/11 12:44, Henning Westerholt escribi?:
> Hi Anto,
>
> having different systems is one solution, you could also just use internally
> in one server different traffic classes for the user sets you have.
> You should also look into the htable module for account/ password brute force
> detection. If you want to implement something w/r to the origination country
> of certain traffic, there is the geoip module which can provide you this
> informations, wich you could also combine with the modules already discussed.
> With regards to user accounts, one thing you probably want to look at is
> password security and fraud or misuse detection logic. Smart attackers will
> find ways around your proactive measures, and you need to respond actively,
> like blocking users after you detected something malicious.
>
> Best regards,
>
> Henning
Hi

There were planning to use the module htable and pike to prevent 
attacks. The biggest question I have is: if I have an account and 
simultaneously make a call would be nice, but I have another account to 
send multiple simultaneous calls, the pike module might detect this as 
"aggressive" and detect a false positive, if configured a level for the 
first, nor is it viable proxy depending on several channels. Could be 
allocated in usr_preference in number of channels, but you might not 
know ... I can not think how.

Another option is to store a variable login attempts and block the 
account X minutes. Thank you very much.

Regards
Anto


From miconda at gmail.com  Sat Jun 18 23:35:32 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Sat, 18 Jun 2011 23:35:32 +0200
Subject: [SR-Users] 100rel without direct reachability
In-Reply-To: <4DFD12FE.2090605@evaristesys.com>
References: <4DFD12FE.2090605@evaristesys.com>
Message-ID: <4DFD1A24.20700@gmail.com>

Hello,


On 6/18/11 11:05 PM, Alex Balashov wrote:
> (Sorry for cross-posting from Sip-Implementors;  however, upon some 
> reflection this is really more of a Kamailio mechanics question.)
>
> Greetings,
>
> I have a scenario where endpoint A is calling endpoint B through proxy
> P.  Endpoints A and B are on different networks that are not directly
> reachable--A is on the public Internet, to which proxy P is connected,
> but NAT'd, and B is on a private RFC1918 network on which the proxy P
> has an interface as well.
>
> The purpose of proxy P is to make calling A <-> B possible, while
> providing the appropriate far-end NAT traversal functionality for A
> and so on and so on.  It does this by inserting the correct
> Record-Route headers into the INVITE from A, and otherwise by staying
> in the signaling path;  both A and B can reach the proxy, so all
> requests and replies go through it.
>
> The problem I'm having is prior to full dialog establishment.  Upon
> receipt of INVITE from A by way of proxy P, B replies with 183 + SDP
> and also:
>
>    Require: 100rel
>
> In the 183 it provides a Contact on its private network that A
> obviously cannot reach:
>
>    Contact: <sip:xxxxx at 192.168.0.10:5060>
>
> As a result, A attempts to send PRACKs to sip:xxxxx at 192.168.0.10,
> which are obviously not going to go anywhere.  After not receiving a
> few of these expected PRACKs, B replies to the call with a final
> negative failure.
>
> The question is, is there anything I can twiddle on the proxy to make
> this work?  The problem with PRACKs is that although they are
> sequential requests, they arise (in this case) prior to the point at
> which RR can be used to guide them through the proxy.
>
> The most obvious answer is, "Make B not Require: 100rel", of course.
> However, that is not an option here.
iirc, the PRACK should follow the record-routing as part of its early 
dialog state. At least, from my experience the PRACKs I saw so far were 
sent to the proxy address due to record routing, not directly to contact 
address in 183. Without following the record routing, I see no way to go 
even for cases with NAT. For me looks like A is buggy, but I haven't 
checked actually the specs at this time.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From abalashov at evaristesys.com  Sat Jun 18 23:59:58 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sat, 18 Jun 2011 17:59:58 -0400
Subject: [SR-Users] 100rel without direct reachability
In-Reply-To: <4DFD1A24.20700@gmail.com>
References: <4DFD12FE.2090605@evaristesys.com> <4DFD1A24.20700@gmail.com>
Message-ID: <F7DF213B-C9AC-40D5-89F9-CAD5E6AEAE47@evaristesys.com>

On Jun 18, 2011, at 5:35 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> iirc, the PRACK should follow the record-routing as part of its early dialog state. At least, from my experience the PRACKs I saw so far were sent to the proxy address due to record routing, not directly to contact address in 183. Without following the record routing, I see no way to go even for cases with NAT. For me looks like A is buggy, but I haven't checked actually the specs at this time.
> 

Daniel,

Sorry, perhaps I was not clear.  A is a softphone initiating a call through Kamailio.  Kamailio forwards the call to UAS B with Record-Route header added, and UAS B sends 183s back to A through Kamailio, with 100rel Require'd.  There are no RR headers seen by A, and so it does not know to send PRACK to something but the Contact in the 183.

From abalashov at evaristesys.com  Sun Jun 19 00:03:40 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sat, 18 Jun 2011 18:03:40 -0400
Subject: [SR-Users] 100rel without direct reachability
In-Reply-To: <F7DF213B-C9AC-40D5-89F9-CAD5E6AEAE47@evaristesys.com>
References: <4DFD12FE.2090605@evaristesys.com> <4DFD1A24.20700@gmail.com>
	<F7DF213B-C9AC-40D5-89F9-CAD5E6AEAE47@evaristesys.com>
Message-ID: <4465DAA4-F4E2-4041-A26B-15C8A26B9D95@evaristesys.com>

My understanding is that the deficiency in the standard here is that B is not required to copy RR headers into provisional responses, only final ones.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jun 18, 2011, at 5:59 PM, Alex Balashov <abalashov at evaristesys.com> wrote:

> On Jun 18, 2011, at 5:35 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
> 
>> iirc, the PRACK should follow the record-routing as part of its early dialog state. At least, from my experience the PRACKs I saw so far were sent to the proxy address due to record routing, not directly to contact address in 183. Without following the record routing, I see no way to go even for cases with NAT. For me looks like A is buggy, but I haven't checked actually the specs at this time.
>> 
> 
> Daniel,
> 
> Sorry, perhaps I was not clear.  A is a softphone initiating a call through Kamailio.  Kamailio forwards the call to UAS B with Record-Route header added, and UAS B sends 183s back to A through Kamailio, with 100rel Require'd.  There are no RR headers seen by A, and so it does not know to send PRACK to something but the Contact in the 183.
> _______________________________________________
> 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  Sun Jun 19 00:17:40 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Sun, 19 Jun 2011 00:17:40 +0200
Subject: [SR-Users] 100rel without direct reachability
In-Reply-To: <4465DAA4-F4E2-4041-A26B-15C8A26B9D95@evaristesys.com>
References: <4DFD12FE.2090605@evaristesys.com> <4DFD1A24.20700@gmail.com>
	<F7DF213B-C9AC-40D5-89F9-CAD5E6AEAE47@evaristesys.com>
	<4465DAA4-F4E2-4041-A26B-15C8A26B9D95@evaristesys.com>
Message-ID: <4DFD2404.4020003@gmail.com>



On 6/19/11 12:03 AM, Alex Balashov wrote:
> My understanding is that the deficiency in the standard here is that B is not required to copy RR headers into provisional responses, only final ones.
indeed, if 183 does not copy record-route headers is not much to do in A 
side -- I expected that any 1xx reply apart of 100, and especially 183 
to mirror record-route headers. I searched a bit and in several cases I 
could spot, the 183 had the R-R headers.

So the fault would be now in B, but if there is no _MUST_ for 183 to 
mirror R-R, then it is in IETF WG :-) ...

Maybe you can fix it via Kamailio configuration file:
- if there is no R-R header in 183 and you know it is going to be a 
bridged call (via some flag set or a special onreply route), then insert 
the headers manually with append_hf(R-R: <sip:ip1;lr>\r\nR-R: 
<sip:ip2;lr>\r\n");

Then see if A is following the R-R.

Cheers,
Daniel
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Jun 18, 2011, at 5:59 PM, Alex Balashov<abalashov at evaristesys.com>  wrote:
>
>> On Jun 18, 2011, at 5:35 PM, Daniel-Constantin Mierla<miconda at gmail.com>  wrote:
>>
>>> iirc, the PRACK should follow the record-routing as part of its early dialog state. At least, from my experience the PRACKs I saw so far were sent to the proxy address due to record routing, not directly to contact address in 183. Without following the record routing, I see no way to go even for cases with NAT. For me looks like A is buggy, but I haven't checked actually the specs at this time.
>>>
>> Daniel,
>>
>> Sorry, perhaps I was not clear.  A is a softphone initiating a call through Kamailio.  Kamailio forwards the call to UAS B with Record-Route header added, and UAS B sends 183s back to A through Kamailio, with 100rel Require'd.  There are no RR headers seen by A, and so it does not know to send PRACK to something but the Contact in the 183.
>> _______________________________________________
>> 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://linkedin.com/in/miconda -- http://twitter.com/miconda



From abalashov at evaristesys.com  Sun Jun 19 00:34:13 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sat, 18 Jun 2011 18:34:13 -0400
Subject: [SR-Users] 100rel without direct reachability
In-Reply-To: <4DFD2404.4020003@gmail.com>
References: <4DFD12FE.2090605@evaristesys.com> <4DFD1A24.20700@gmail.com>
	<F7DF213B-C9AC-40D5-89F9-CAD5E6AEAE47@evaristesys.com>
	<4465DAA4-F4E2-4041-A26B-15C8A26B9D95@evaristesys.com>
	<4DFD2404.4020003@gmail.com>
Message-ID: <4DFD27E5.8030306@evaristesys.com>

On 06/18/2011 06:17 PM, Daniel-Constantin Mierla wrote:

> indeed, if 183 does not copy record-route headers is not much to do in
> A side -- I expected that any 1xx reply apart of 100, and especially
> 183 to mirror record-route headers. I searched a bit and in several
> cases I could spot, the 183 had the R-R headers.
>
> So the fault would be now in B, but if there is no _MUST_ for 183 to
> mirror R-R, then it is in IETF WG :-) ...

Well, here's what 3261 12.1.1 ("UAS Behavior") says on the subject:

    When a UAS responds to a request with a response that establishes
    a dialog (such as a 2xx to INVITE), the UAS MUST copy all
    Record-Route header field values from the request into the
    response (including the URIs, URI parameters, and any
    Record-Route header field parameters, whether they are known
    or unknown to the UAS) and MUST maintain the order of those
    values.

Since a 183 does not establish a dialog, I suppose that means there's 
no need for the UAS to copy the RR headers.

On the other hand, take a look at page 7 of RFC 3262:

    http://www.ietf.org/rfc/rfc3262.txt

It seems to say that Record-Route goes into "2xx,18x", although offers 
no elaboration on this point.  Whether this constitutes an amendment 
to the 3261 view I do not know.

> - if there is no R-R header in 183 and you know it is going to be a
> bridged call (via some flag set or a special onreply route), then
> insert the headers manually with append_hf(R-R: <sip:ip1;lr>\r\nR-R:
> <sip:ip2;lr>\r\n");

Yeah, that was going to be my next step, I just figured I'd try to 
avoid something so ugly.  :-)

Thanks for your help!

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From ibc at aliax.net  Sun Jun 19 14:50:40 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sun, 19 Jun 2011 14:50:40 +0200
Subject: [SR-Users] 100rel without direct reachability
In-Reply-To: <4DFD27E5.8030306@evaristesys.com>
References: <4DFD12FE.2090605@evaristesys.com> <4DFD1A24.20700@gmail.com>
	<F7DF213B-C9AC-40D5-89F9-CAD5E6AEAE47@evaristesys.com>
	<4465DAA4-F4E2-4041-A26B-15C8A26B9D95@evaristesys.com>
	<4DFD2404.4020003@gmail.com> <4DFD27E5.8030306@evaristesys.com>
Message-ID: <BANLkTimvC-hG9UZfB6__h+QuPceOMNR+uA@mail.gmail.com>

2011/6/19 Alex Balashov <abalashov at evaristesys.com>:
> Well, here's what 3261 12.1.1 ("UAS Behavior") says on the subject:
>
> ? When a UAS responds to a request with a response that establishes
> ? a dialog (such as a 2xx to INVITE), the UAS MUST copy all
> ? Record-Route header field values from the request into the
> ? response (including the URIs, URI parameters, and any
> ? Record-Route header field parameters, whether they are known
> ? or unknown to the UAS) and MUST maintain the order of those
> ? values.
>
> Since a 183 does not establish a dialog, I suppose that means there's no
> need for the UAS to copy the RR headers.

Wrong, 183 *does* establish a dialog. An early-dialog in fact, but
that doesn't matter. As said above, 1XX response with require 100rel
is a reliable response so it requires Contact header and mirroring
Record-Route headers.


> On the other hand, take a look at page 7 of RFC 3262:
>
> ? http://www.ietf.org/rfc/rfc3262.txt
>
> It seems to say that Record-Route goes into "2xx,18x", although offers no
> elaboration on this point. ?Whether this constitutes an amendment to the
> 3261 view I do not know.

It says:


            Header field              where      PRACK
            __________________________________________
            Record-Route                R          o
            Record-Route             2xx,18x       o


Of course Record-Route is optional in request (INVITE) and 18X or 2XX
responses, as the proxy could decide not to do record-routing.

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


From ibc at aliax.net  Sun Jun 19 20:42:24 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sun, 19 Jun 2011 20:42:24 +0200
Subject: [SR-Users] Security in proxy
In-Reply-To: <4DF93B3C.9060109@gmail.com>
References: <4DF93B3C.9060109@gmail.com>
Message-ID: <BANLkTimnK4_RG46D8Nn+b4gEn1k53gd-vw@mail.gmail.com>

2011/6/16 Anto <potxoka at gmail.com>:
> Is there any way to implement something that detects unusual behavior on the
> user (referred to countries that do not ever called, excessive traffic,
> etc)?.

This requirement is too complex to implement it in a generic module
(IMHO). For example, if we are a Spanish telco operator and our
clients are mostly Spanish, they usually call Spanish numbers. If one
of them starts making lot of calls to Slovenia it could be an attack.
But maybe we have a client that, in fact, makes lot of calls to
Slovenia por any valid reason (businness model and so). So each client
could require a different "security profile".

We are planning to build a system for this purpose. The idea is having
a separate custom server (not a SIP server, or maybe) that collects
clients calls and analyze them against a "security profiles backend"
(a DB or whatever). When kamailio receives a call it would communicate
with this server in some way and ask it "client XXXX wants to make a
call tu NNNNNNN, should I allow it?". Then the custom server would
apply internal logic and give a fast response (yes/no).

I think this is too complex and could make sense to build a
separate/custom server for this purpose, and then, maybe create a
custom Kamailio module for communicating Kamailio with the server
(some communication protocol and so).


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


From ibc at aliax.net  Sun Jun 19 20:49:24 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sun, 19 Jun 2011 20:49:24 +0200
Subject: [SR-Users] Invite replies
In-Reply-To: <BANLkTinocH0337wuSxCWTVyAY5a52NT+MQ@mail.gmail.com>
References: <BANLkTinocH0337wuSxCWTVyAY5a52NT+MQ@mail.gmail.com>
Message-ID: <BANLkTimXdc5aR3fPCpsYCX56Y4dzePgy1w@mail.gmail.com>

2011/6/16 Lucas Alvarez <lucasaa at gmail.com>:
> Hi, how may I catch the reply of an invite? I did this but it seems is not
> working:
> ?if (method=="INVITE") {
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?$avp(sdp_ret) = fix_nated_sdp("11");
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?xlog("L_ALERT","#####********###### SDP
> return: $avp(sdp_ret)\n");
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?append_hf("P-hint: sdp rewritten\r\n");
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?t_on_reply("REPLY_TWO");
> }
> onreply_route("REPLY_TWO") {
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?xlog("L_ALERT","ALERT: ################ ?$rU
> - $rm from $fu (IP:$si:$sp--------$di)\n");
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?return;
> }
>
> Any help will be?appreciate it, thanks in advance.


Where is the t_relay() call?
And, which is the problem? does the reply arrives to kamailio or not?


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

From ggb at tid.es  Mon Jun 20 00:15:34 2011
From: ggb at tid.es (Gustavo Garcia Bernardo)
Date: Mon, 20 Jun 2011 00:15:34 +0200
Subject: [SR-Users] Keeping transactions in standby till future events
Message-ID: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>

Is it possible to configure sip-router to keep a transaction for an incoming message (f.e. an INVITE) without sending a response and without forwarding it, and then reference and complete it in the future?

Using pseudocode this is the logic I would like to implement:

//When receiving an INVITE leave it pending
if (request is INVITE) {
    store transactionID  // This transactionID will be send somehow to the client
    return
}

//When receiving a PUBLISH referencing a pending transaction complete the forwarding
if (request is PUBLISH) {
    transactionId = header("X-TransactionId")
    destination = header("X-Destination")
    transactions [transactionID]. forward(destination)
}

Thank you very much,
G.

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This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at.
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From abalashov at evaristesys.com  Mon Jun 20 00:20:05 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sun, 19 Jun 2011 18:20:05 -0400
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
Message-ID: <4DFE7615.4040606@evaristesys.com>

There is no support for this.  Kamailio in its proxy role (the role 
you are invoking) is pretty much event-driven, in the sense that it's 
processing is real-time and correlated to the receipt of messages to 
be forwarded.  It is externally stimulated into action.

Also, it would be quite problematic from the perspective of SIP 
protocol mechanics.  If you do not even send 100 Trying, the sending 
UAC will retransmit the INVITE, and/or eventually assume your proxy is 
down.  If you do send 100 Trying but do not follow up, other timers 
awaiting final positive or negative replies kick in and also expire.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From henning.westerholt at 1und1.de  Mon Jun 20 11:06:14 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Mon, 20 Jun 2011 11:06:14 +0200
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
Message-ID: <201106201106.14556.henning.westerholt@1und1.de>

On Monday 20 June 2011, Gustavo Garcia Bernardo wrote:
> Is it possible to configure sip-router to keep a transaction for an
> incoming message (f.e. an INVITE) without sending a response and without
> forwarding it, and then reference and complete it in the future?
> 
> Using pseudocode this is the logic I would like to implement:
> [..]

Hi Gustavo,

as Alex already wrote, its probably not the best or correct thing to do - but 
have you already looked to the tm module? It provides some asynchronous 
support. From the module README:

"[t_supend] This function together with t_continue() can be used to implement 
asynchronous actions: t_suspend() saves the transaction, returns its 
identifiers, and t_continue() continues the SIP request processing."

Maybe this is helpful,

Henning


From henning.westerholt at 1und1.de  Mon Jun 20 11:11:11 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Mon, 20 Jun 2011 11:11:11 +0200
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
In-Reply-To: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>
References: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>
Message-ID: <201106201111.11708.henning.westerholt@1und1.de>

On Saturday 18 June 2011, Omar wrote:
> I am Using Solaris 10 and GCC 4.2, i do have libconfuse
> 
> everything goes fine until this error comes up.
> 
> carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a
> function) carrierroute.c:87: error: expected '=', ',', ';', 'asm' or
> '__attribute__' before 'cmds' carrierroute.c:132: error: variable
> 'exports' has initializer but incomplete type carrierroute.c:133: warning:
> excess elements in struct initializer carrierroute.c:133: warning: (near
> initialization for 'exports')
> carrierroute.c:134: warning: excess elements in struct initializer
> carrierroute.c:134: warning: (near initialization for 'exports')
> [..]

Hi Omar,

the error messages looks not really related to carrierroute module 
dependencies for me - but its worth to check anyway: Do you've also the 
libconfuse development headers installed? Other modules compile just fine?

Best regards,

Henning


From abalashov at evaristesys.com  Mon Jun 20 11:21:06 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 20 Jun 2011 05:21:06 -0400
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <201106201106.14556.henning.westerholt@1und1.de>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
	<201106201106.14556.henning.westerholt@1und1.de>
Message-ID: <4DFF1102.3010005@evaristesys.com>

Hi Henning,

On 06/20/2011 05:06 AM, Henning Westerholt wrote:

> "[t_supend] This function together with t_continue() can be used to implement
> asynchronous actions: t_suspend() saves the transaction, returns its
> identifiers, and t_continue() continues the SIP request processing."

These are just exported API functions though, right?  Not something 
that can be called from script?

Thanks,

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From henning.westerholt at 1und1.de  Mon Jun 20 11:39:32 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Mon, 20 Jun 2011 11:39:32 +0200
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <4DFF1102.3010005@evaristesys.com>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
	<201106201106.14556.henning.westerholt@1und1.de>
	<4DFF1102.3010005@evaristesys.com>
Message-ID: <201106201139.32706.henning.westerholt@1und1.de>

On Monday 20 June 2011, Alex Balashov wrote:
> > "[t_supend] This function together with t_continue() can be used to
> > implement asynchronous actions: t_suspend() saves the transaction,
> > returns its identifiers, and t_continue() continues the SIP request
> > processing."
> 
> These are just exported API functions though, right?  Not something
> that can be called from script?

Hi Alex,

yes, they are in the available in the public tm API:

int t_suspend(struct sip_msg *msg,
             unsigned int *hash_index, unsigned int *label)

int t_continue(unsigned int hash_index, unsigned int label,
             struct action *route)

Best regards,

Henning


From ggb at tid.es  Mon Jun 20 13:41:16 2011
From: ggb at tid.es (Gustavo Garcia Bernardo)
Date: Mon, 20 Jun 2011 13:41:16 +0200
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <201106201139.32706.henning.westerholt@1und1.de>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
	<201106201106.14556.henning.westerholt@1und1.de>
	<4DFF1102.3010005@evaristesys.com>
	<201106201139.32706.henning.westerholt@1und1.de>
Message-ID: <B348B152E5F11640B2247E54304E53FC2672B1776F@EXCLU2K7.hi.inet>

Thank you, that is the functionality we needed but we were looking for something to be accessible from the script.   Perhaps we can build a module to export those APIs to the script, we will try it.

Best regards,
G.

-----Original Message-----
From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Henning Westerholt
Sent: lunes, 20 de junio de 2011 11:40
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Keeping transactions in standby till future events

On Monday 20 June 2011, Alex Balashov wrote:
> > "[t_supend] This function together with t_continue() can be used to
> > implement asynchronous actions: t_suspend() saves the transaction,
> > returns its identifiers, and t_continue() continues the SIP request
> > processing."
>
> These are just exported API functions though, right?  Not something
> that can be called from script?

Hi Alex,

yes, they are in the available in the public tm API:

int t_suspend(struct sip_msg *msg,
             unsigned int *hash_index, unsigned int *label)

int t_continue(unsigned int hash_index, unsigned int label,
             struct action *route)

Best regards,

Henning

_______________________________________________
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

Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo.
This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at.
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From s.felici at mclink.eu  Mon Jun 20 14:50:10 2011
From: s.felici at mclink.eu (Simone Felici)
Date: Mon, 20 Jun 2011 14:50:10 +0200
Subject: [SR-Users] Help find out where to modify request uri
In-Reply-To: <4DFB6975.3000404@mclink.eu>
References: <4DFB2357.60802@mclink.eu> <4DFB5952.7090705@gmail.com>
	<4DFB6975.3000404@mclink.eu>
Message-ID: <4DFF4202.1040607@mclink.eu>


Hi again,

I've found where I can collect all infromations i need  in the "INBOUND" route (called username and 
alias). I've saved them into two variables that I've printed out to log to confirm. Into NAT_MANGLE 
route I can match the uri that I should change, based on a username:

if (uri=~"^sip:MYUSERNAME@")  {
    xlog("L_ERR", "LXLOG MATCH!!!! - %$alias_number - %$user_number");
}

It works. But I've no idea how now change the uri to use the new alias instead the old one.
I.ex from original:

<sip:<MYUSERNAME>@<IP>:1032;uniq=7600673C6E68DF3228C4E238ABB24>

to

<sip:<NEWUSERNAME>@<IP>:1032;uniq=7600673C6E68DF3228C4E238ABB24>

When I'll able to edit it statically, I'll then try db_ops to change values based on some MySQL 
entries. For now it would great if I could simple edit the uri on a static example.

It's not so clear how to use the t_relay or whatever I could use to rewrite my new destination based 
on the above needs.

Any help?

Thank's a lot!

Simon






Il 17/06/2011 16:49, Simone Felici ha scritto:
>
> Thank you for your answer.
> You've absolutely right on the linked ser.cfg. I send a cleaned up and indented version.
> It's attached.
> Meanwhile I'll try out a little your solution.
>
> Thank's
>
> Simon


From ibc at aliax.net  Mon Jun 20 15:29:42 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 20 Jun 2011 15:29:42 +0200
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <201106201106.14556.henning.westerholt@1und1.de>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
	<201106201106.14556.henning.westerholt@1und1.de>
Message-ID: <BANLkTikybaL_3x+yx_czFLARU8t3i3kGdA@mail.gmail.com>

2011/6/20 Henning Westerholt <henning.westerholt at 1und1.de>:
> as Alex already wrote, its probably not the best or correct thing to do - but
> have you already looked to the tm module? It provides some asynchronous
> support. From the module README:
>
> "[t_supend] This function together with t_continue() can be used to implement
> asynchronous actions: t_suspend() saves the transaction, returns its
> identifiers, and t_continue() continues the SIP request processing."

Hi Henning, which would be the difference if the script does
t_newtran(), then reply 100 trying (of before) and then exit()?

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


From s.felici at mclink.eu  Mon Jun 20 15:36:09 2011
From: s.felici at mclink.eu (Simone Felici)
Date: Mon, 20 Jun 2011 15:36:09 +0200
Subject: [SR-Users] Help find out where to modify request uri
In-Reply-To: <4DFF4202.1040607@mclink.eu>
References: <4DFB2357.60802@mclink.eu>
	<4DFB5952.7090705@gmail.com>	<4DFB6975.3000404@mclink.eu>
	<4DFF4202.1040607@mclink.eu>
Message-ID: <4DFF4CC9.2090901@mclink.eu>

Sorry again!

I've found how to change it but I'm missing something maybe stupid.
Having the new number to call into a variable $alias_number I can print it out as follow:

xlog("L_ERR", "MATCH USERNAME - %$alias_number - %$user_number");

I've tried to rewrite it into the if-statement using both examples:

rewriteuser("$alias_number");
OR
subst_uri('/^sip:(.*)@(.*)$/sip:$alias_number@\2/');

But the URI will rewritten with the WORD "$alias_number" instead of it's value. (tried with 
$alias_number as well as %$alias_number).

Where I'm wrong?

Thank's!

Simon




Il 20/06/2011 14:50, Simone Felici ha scritto:
>
> Hi again,
>
> I've found where I can collect all infromations i need in the "INBOUND" route (called username and
> alias). I've saved them into two variables that I've printed out to log to confirm. Into NAT_MANGLE
> route I can match the uri that I should change, based on a username:
>
> if (uri=~"^sip:MYUSERNAME@") {
> xlog("L_ERR", "LXLOG MATCH!!!! - %$alias_number - %$user_number");
> }
>
> It works. But I've no idea how now change the uri to use the new alias instead the old one.
> I.ex from original:
>
> <sip:<MYUSERNAME>@<IP>:1032;uniq=7600673C6E68DF3228C4E238ABB24>
>
> to
>
> <sip:<NEWUSERNAME>@<IP>:1032;uniq=7600673C6E68DF3228C4E238ABB24>
>
> When I'll able to edit it statically, I'll then try db_ops to change values based on some MySQL
> entries. For now it would great if I could simple edit the uri on a static example.
>
> It's not so clear how to use the t_relay or whatever I could use to rewrite my new destination based
> on the above needs.
>
> Any help?
>
> Thank's a lot!
>
> Simon
>
>
>
>
>
>
> Il 17/06/2011 16:49, Simone Felici ha scritto:
>>
>> Thank you for your answer.
>> You've absolutely right on the linked ser.cfg. I send a cleaned up and indented version.
>> It's attached.
>> Meanwhile I'll try out a little your solution.
>>
>> Thank's
>>
>> 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


-- 
Simone Felici
Divisione Tecnica: Progettazione e Sviluppo

tel. +39 0461.030.111
fax. +39 0461 030.112
Via Fersina, 23 - 38123 Trento

-------------
MC-link S.p.A.
Sede Direzionale e Amministrativa
Via Carlo Perrier, 9/a - 00157 Roma
Sede Legale
Via Fersina, 23 - 38123 Trento

http://www.mclink.it

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From miconda at gmail.com  Mon Jun 20 15:41:19 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 20 Jun 2011 15:41:19 +0200
Subject: [SR-Users] The 15000th development commit
Message-ID: <4DFF4DFF.1070602@gmail.com>

Hello,

I just wanted to mark the 15000th development commit in our project 
(counted only on GIT master branch). It happened several days ago and it 
was done by the same person that did the first commit, Andrei:

git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1

512dcd9    Andrei Pelinescu-Onciul    Mon Sep 3 21:27:11 2001 +0000    
Initial revision

The 15000th one is:

git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 
| tail -1

8a90dd3    Andrei Pelinescu-Onciul    Sat Jun 11 11:24:05 2011 +0200    
core: remove unused variables + coding style


I published the news about it on project sites as well:
   * http://asipto.com/u/37
   * http://asipto.com/u/38

Now it's the time to look forward to the 20000th commit!

Enjoy the summer, autumn comes with v3.2.0,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From omar at 321communications.com  Mon Jun 20 15:47:52 2011
From: omar at 321communications.com (Omar)
Date: Mon, 20 Jun 2011 09:47:52 -0400
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
In-Reply-To: <201106201111.11708.henning.westerholt@1und1.de>
References: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>
	<201106201111.11708.henning.westerholt@1und1.de>
Message-ID: <8869B5C4-D4D4-4B06-9015-7E753BC68E6C@321communications.com>

Thank you for answer Henning

Everything compiles fine until the showed error the were errors about the libconfuse which I fixed stating the path to the library, yes lib confuse is installed I am using the same one I installed for the kamailio 1.4.3.



Omar

On Jun 20, 2011, at 5:11 AM, Henning Westerholt <henning.westerholt at 1und1.de> wrote:

> On Saturday 18 June 2011, Omar wrote:
>> I am Using Solaris 10 and GCC 4.2, i do have libconfuse
>> 
>> everything goes fine until this error comes up.
>> 
>> carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a
>> function) carrierroute.c:87: error: expected '=', ',', ';', 'asm' or
>> '__attribute__' before 'cmds' carrierroute.c:132: error: variable
>> 'exports' has initializer but incomplete type carrierroute.c:133: warning:
>> excess elements in struct initializer carrierroute.c:133: warning: (near
>> initialization for 'exports')
>> carrierroute.c:134: warning: excess elements in struct initializer
>> carrierroute.c:134: warning: (near initialization for 'exports')
>> [..]
> 
> Hi Omar,
> 
> the error messages looks not really related to carrierroute module 
> dependencies for me - but its worth to check anyway: Do you've also the 
> libconfuse development headers installed? Other modules compile just fine?
> 
> Best regards,
> 
> Henning
> 


From ibc at aliax.net  Mon Jun 20 16:08:02 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 20 Jun 2011 16:08:02 +0200
Subject: [SR-Users] Keeping transactions in standby till future events
In-Reply-To: <BANLkTikybaL_3x+yx_czFLARU8t3i3kGdA@mail.gmail.com>
References: <B348B152E5F11640B2247E54304E53FC2672606546@EXCLU2K7.hi.inet>
	<201106201106.14556.henning.westerholt@1und1.de>
	<BANLkTikybaL_3x+yx_czFLARU8t3i3kGdA@mail.gmail.com>
Message-ID: <BANLkTi=TgEgDBcob1Tzo1Cpi9-c6JxZLvA@mail.gmail.com>

2011/6/20 I?aki Baz Castillo <ibc at aliax.net>:
>> "[t_supend] This function together with t_continue() can be used to implement
>> asynchronous actions: t_suspend() saves the transaction, returns its
>> identifiers, and t_continue() continues the SIP request processing."
>
> Hi Henning, which would be the difference if the script does
> t_newtran(), then reply 100 trying (of before) and then exit()?

Auto reply: I think it would just generate a "ready-to-expire" server
transaction.

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


From vladiksip at centrum.cz  Mon Jun 20 16:32:04 2011
From: vladiksip at centrum.cz (Vladimir Broz)
Date: Mon, 20 Jun 2011 16:32:04 +0200
Subject: [SR-Users] an IDEA
In-Reply-To: <BANLkTiktPJ4Oyzb6isep8ZSL03-Co3BHaw@mail.gmail.com>
References: <BANLkTiktPJ4Oyzb6isep8ZSL03-Co3BHaw@mail.gmail.com>
Message-ID: <4DFF59E4.8020106@centrum.cz>

Hi,

I think you will need a SIP B2BUA - check SEMS - http://www.iptel.org/sems.

Hope that helps,
-Vlada

On 06/18/2011 03:25 PM, afshin afzali wrote:
> Hi Guys,
>
> I need to integrate a SIP processing environment to H.248 and SS7/ ISUP
> links.
> For those communication links, OpenSS7 and Erlang/OTP MEGACO are good
> solutions. My idea is to write applications over OTP and SIP-Router to
> be able to bridge non-SIP protocols to SIP!
> On OTP the way is clear, but on SIP-Router I need to be able to generate
> / terminate SIP transactions. I'll appreciate every comment.
>
> BEST,
> -- afshin
>
>
>
> _______________________________________________
> 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 potxoka at gmail.com  Mon Jun 20 17:04:25 2011
From: potxoka at gmail.com (Anto)
Date: Mon, 20 Jun 2011 17:04:25 +0200
Subject: [SR-Users] Security in proxy
In-Reply-To: <4DF93B3C.9060109@gmail.com>
References: <4DF93B3C.9060109@gmail.com>
Message-ID: <4DFF6179.3060500@gmail.com>

Hi

Thinking about what to have multiple proxy according to their use would 
have to have different gateways for each proxy, right?. If you have the 
same gateway for different types of use of proxy, the gateway will have 
to look for so that proxy is the incoming traffic and send it. You might 
also have the same gateway and be a proxy (balancer) to redirect traffic 
to the corresponding proxy. Or have a policy of different traffic input 
to output. Thank you.

Regards
Anto


From mailshariftanvir at gmail.com  Mon Jun 20 15:23:45 2011
From: mailshariftanvir at gmail.com (Sharif Tanvir Rahman)
Date: Mon, 20 Jun 2011 19:23:45 +0600
Subject: [SR-Users] Lua script calling rewriteuri
Message-ID: <BANLkTimcdd4F=5qiJ9gYiod5RjTVPYMuGg@mail.gmail.com>

Hello,

We plan to use Kamailio as redirect server. Corresponding contact list is to
be fetched by a Lua script from DB.

The problem is about calling the required 'rewriteuri' function:

   1. From lua script. both 'sr.rewriteuri()' and 'sr.modf("rewriteuri",
   ...)" fails to write the 'contact' field.
   2. I tried an alternative way; pushing the value from lua (e.g.
   sr.pv.sets("contact", "sip:...")) and got the value in the Kamailio.cfg
   ($var(contact)). But can't call the rewriteuri function with this variable's
   value as argument. This seems to be a general problem of calling a function
   from .cfg with a variable's value as parameter.

Any help much appreciated.

-- 
Sharif
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From tayeb.meftah at gmail.com  Mon Jun 20 15:37:19 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 20 Jun 2011 15:37:19 +0200
Subject: [SR-Users] [Kamailio-Business] The 15000th development commit
References: <4DFF4DFF.1070602@gmail.com>
Message-ID: <AB4F45F892DD44FF8E46CA78F03BCE8E@work>

congratulation Andrei !
20000TH would by you, daniel ;)

----- Original Message ----- 
From: "Daniel-Constantin Mierla" <miconda at gmail.com>
To: "kamailio" <sr-users at lists.sip-router.org>; "sr-dev" 
<sr-dev at lists.sip-router.org>; <business at lists.kamailio.org>
Sent: Monday, June 20, 2011 3:41 PM
Subject: [Kamailio-Business] The 15000th development commit


> Hello,
>
> I just wanted to mark the 15000th development commit in our project 
> (counted only on GIT master branch). It happened several days ago and it 
> was done by the same person that did the first commit, Andrei:
>
> git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1
>
> 512dcd9    Andrei Pelinescu-Onciul    Mon Sep 3 21:27:11 2001 +0000 
> Initial revision
>
> The 15000th one is:
>
> git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 | 
> tail -1
>
> 8a90dd3    Andrei Pelinescu-Onciul    Sat Jun 11 11:24:05 2011 +0200 
> core: remove unused variables + coding style
>
>
> I published the news about it on project sites as well:
>   * http://asipto.com/u/37
>   * http://asipto.com/u/38
>
> Now it's the time to look forward to the 20000th commit!
>
> Enjoy the summer, autumn comes with v3.2.0,
> Daniel
>
> -- 
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
> _______________________________________________
> Kamailio (OpenSER) - Business mailing list
> Business at lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/business
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/business
>
>
> __________ Information provenant d'ESET NOD32 Antivirus, version de la 
> base des signatures de virus 6222 (20110620) __________
>
> Le message a ?t? v?rifi? par ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> 


__________ Information provenant d'ESET NOD32 Antivirus, version de la base des signatures de virus 6223 (20110620) __________

Le message a ?t? v?rifi? par ESET NOD32 Antivirus.

http://www.eset.com





From miconda at gmail.com  Tue Jun 21 10:21:25 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 10:21:25 +0200
Subject: [SR-Users] Lua script calling rewriteuri
In-Reply-To: <BANLkTimcdd4F=5qiJ9gYiod5RjTVPYMuGg@mail.gmail.com>
References: <BANLkTimcdd4F=5qiJ9gYiod5RjTVPYMuGg@mail.gmail.com>
Message-ID: <4E005485.3080500@gmail.com>

Hello,

to rewrite the r-uri you can use the function for setting PV value:

sr.pv.sets("$ru", "_here_is_new_uri_");

To add contact headers to reply, you should use:

sr.hdr.append_to_reply("Contact: <_here_is_your_uri_>\r\n");

Cheers,
Daniel

On 6/20/11 3:23 PM, Sharif Tanvir Rahman wrote:
> Hello,
>
> We plan to use Kamailio as redirect server. Corresponding contact list 
> is to be fetched by a Lua script from DB.
>
> The problem is about calling the required 'rewriteuri' function:
>
>  1. From lua script. both 'sr.rewriteuri()' and 'sr.modf("rewriteuri",
>     ...)" fails to write the 'contact' field.
>  2. I tried an alternative way; pushing the value from lua (e.g.
>     sr.pv.sets("contact", "sip:...")) and got the value in the
>     Kamailio.cfg ($var(contact)). But can't call the rewriteuri
>     function with this variable's value as argument. This seems to be
>     a general problem of calling a function from .cfg with a
>     variable's value as parameter.
>
> Any help much appreciated.
>
> -- 
> Sharif
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From miconda at gmail.com  Tue Jun 21 10:27:19 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 10:27:19 +0200
Subject: [SR-Users] [Kamailio-Business] The 15000th development commit
In-Reply-To: <AB4F45F892DD44FF8E46CA78F03BCE8E@work>
References: <4DFF4DFF.1070602@gmail.com>
	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
Message-ID: <4E0055E7.6030808@gmail.com>



On 6/20/11 3:37 PM, Meftah Tayeb wrote:
> congratulation Andrei !
> 20000TH would by you, daniel ;)
I did the commit no 14892, so the 20k-th is someone else turn :-)

git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -14892 
| tail -1
d4a66da    Daniel-Constantin Mierla    Mon Apr 25 12:23:46 2011 +0200    
kex: added mi cmd clear_statistics

Cheers,
Daniel

>
> ----- Original Message ----- From: "Daniel-Constantin Mierla" 
> <miconda at gmail.com>
> To: "kamailio" <sr-users at lists.sip-router.org>; "sr-dev" 
> <sr-dev at lists.sip-router.org>; <business at lists.kamailio.org>
> Sent: Monday, June 20, 2011 3:41 PM
> Subject: [Kamailio-Business] The 15000th development commit
>
>
>> Hello,
>>
>> I just wanted to mark the 15000th development commit in our project 
>> (counted only on GIT master branch). It happened several days ago and 
>> it was done by the same person that did the first commit, Andrei:
>>
>> git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1
>>
>> 512dcd9    Andrei Pelinescu-Onciul    Mon Sep 3 21:27:11 2001 +0000 
>> Initial revision
>>
>> The 15000th one is:
>>
>> git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head 
>> -15000 | tail -1
>>
>> 8a90dd3    Andrei Pelinescu-Onciul    Sat Jun 11 11:24:05 2011 +0200 
>> core: remove unused variables + coding style
>>
>>
>> I published the news about it on project sites as well:
>>   * http://asipto.com/u/37
>>   * http://asipto.com/u/38
>>
>> Now it's the time to look forward to the 20000th commit!
>>
>> Enjoy the summer, autumn comes with v3.2.0,
>> Daniel
>>
>> -- 
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>> _______________________________________________
>> Kamailio (OpenSER) - Business mailing list
>> Business at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/business
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/business
>>
>>
>> __________ Information provenant d'ESET NOD32 Antivirus, version de 
>> la base des signatures de virus 6222 (20110620) __________
>>
>> Le message a ?t? v?rifi? par ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>
>
> __________ Information provenant d'ESET NOD32 Antivirus, version de la 
> base des signatures de virus 6223 (20110620) __________
>
> Le message a ?t? v?rifi? par ESET NOD32 Antivirus.
>
> http://www.eset.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

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From vaad.fabi at gmail.com  Tue Jun 21 11:03:30 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Tue, 21 Jun 2011 12:03:30 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E0055E7.6030808@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com>
Message-ID: <4E005E62.3090205@gmail.com>

  Hello all,

I want to use realtime architecture kamailio 3.1 + asterisk 1.6.2 
described at 
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb,  
but use it with another softswitch (it also use mysql db for configuration).
If it possible please answer for several questions:

1.    What is "#!define WITH_ASTERISK" option doing ?
         Does it only use for:
         - load uac module
         - read login\passw table info
         - some routing cases
     or there are some else jobs ?
Maybe there is some description for "#!define WITH_ASTERISK" ?


2.    What else fields of asterisk realtime table (except 
username\sippasswd) kamailio uses for integration ?

3.    From integration manual:
         "
         CREATE TABLE version (
         table_name VARCHAR(32) NOT NULL,
         table_version INT UNSIGNED DEFAULT 0 NOT NULL
         );
         INSERT INTO version (table_name, table_version) VALUES 
('sipusers','6');
         "
For what purposes this table used ? How to change it for using with 
another softswitch ?



-- 

vf



From lau.testi at gmail.com  Tue Jun 21 11:39:04 2011
From: lau.testi at gmail.com (laura testi)
Date: Tue, 21 Jun 2011 11:39:04 +0200
Subject: [SR-Users] kamailio 3.1.3 fail to start as a service when load the
	ldap.so module
Message-ID: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>

Hi,
we try to use LDAP module in kamailio, first we try only load the ldap.so
only to see if it can be loaded. When we start from the command line with
either "kamctl start" or directly "kamilio", the kamailio is started
normally with different children. When I try to start as service with
"service kamailio start" or "/etc/init.d/kamailio start". It fails with the
following error in the /var/log/messages:
----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]:
init_tcp: using epoll_lt as the io watch method (auto detected)

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc
[hslot.c:53]: locks array size 512

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth
[auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support
disabled

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
[parse_config.c:251]: file not found: //etc/kamailio/permissions.allow

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
[permissions.c:606]: default allow file (//etc/kamailio/permissions.allow)
not found => empty rule set

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
[parse_config.c:251]: file not found: //etc/kamailio/permissions.deny

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
[permissions.c:615]: default deny file (//etc/kamailio/permissions.deny) not
found => empty rule set

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap
[ldap_mod.c:246]: OpenLDAP - 20343

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_init.c:511]: tls: _init_tls_h:  compiled  with  openssl  version
"OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on,
compression: on

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_init.c:519]: tls: init_tls_h: installed openssl library version
"OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on,
zlib compression: on  compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB
-DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT
-I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall -DMD32_REG_T=int -O2 -g
-pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector
--param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack
-DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT
-DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_init.c:373]: tls: init_tls: disabling compression...

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls
[tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug
detected,  kerberos
support will be disabled...

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls
[tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory)
workaround enabled (on low memory tls operations will fail preemptively)
with free memory thresholds 9961472 and 4980736 bytes

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been
changed to 9961472

Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been
changed to 4980736

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:176]: TLSs<default>: tls_method=9

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:188]: TLSs<default>:
certificate='//etc/kamailio/FBCIDS01V01.pem'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:202]: TLSs<default>: crl='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:206]: TLSs<default>: require_certificate=0

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:213]: TLSs<default>: cipher_list='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:220]: TLSs<default>:
private_key='//etc/kamailio/FBCIDS01V01KEY.pem'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:224]: TLSs<default>: verify_certificate=0

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:227]: TLSs<default>: verify_depth=9

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:544]: TLSs<default>: No client certificate required and no
checks performed

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
workaround is needed

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:176]: TLSc<default>: tls_method=9

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:188]: TLSc<default>: certificate='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:195]: TLSc<default>: ca_list='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:202]: TLSc<default>: crl='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:206]: TLSc<default>: require_certificate=0

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:213]: TLSc<default>: cipher_list='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:220]: TLSc<default>: private_key='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:224]: TLSc<default>: verify_certificate=0

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:227]: TLSc<default>: verify_depth=9

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
workaround is needed

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:176]: TLSs<10.10.5.214:5061>: tls_method=12

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:188]: TLSs<10.10.5.214:5061>:
certificate='//etc/kamailio/FBCIDS01V01.pem'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:195]: TLSs<10.10.5.214:5061>:
ca_list='//etc/kamailio/tiglobalca.pem'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:202]: TLSs<10.10.5.214:5061>: crl='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:206]: TLSs<10.10.5.214:5061>: require_certificate=0

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:213]: TLSs<10.10.5.214:5061>: cipher_list='(null)'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:220]: TLSs<10.10.5.214:5061>:
private_key='//etc/kamailio/FBCIDS01V01KEY.pem'

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:224]: TLSs<10.10.5.214:5061>: verify_certificate=0

Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:227]: TLSs<10.10.5.214:5061>: verify_depth=9

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:544]: TLSs<10.10.5.214:5061>: No client certificate required
and no checks performed

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
[tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
workaround is needed

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
[ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
server

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
[ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core>
[sr_module.c:832]: init_mod_child(): Error while initializing module ldap
(/usr/lib64/kamailio/modules_k/ldap.so)

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core>
[pt.c:337]: ERROR: fork_process(): init_child failed for  process 1, pid
8786, "udp receiver child=0 sock=127.0.0.1:5060"

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core>
[main.c:1546]: main_loop: Cannot fork

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
[ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
server

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
[ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core>
[sr_module.c:832]: init_mod_child(): Error while initializing module ldap
(/usr/lib64/kamailio/modules_k/ldap.so)

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core>
[pt.c:337]: ERROR: fork_process(): init_child failed for  process 2, pid
8788, "udp receiver child=1 sock=127.0.0.1:5060"

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core>
[main.c:1546]: main_loop: Cannot fork

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core>
[main.c:738]: child process 8786 exited normally, status=255

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core>
[main.c:738]: child process 8788 exited normally, status=255

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
[main.c:756]: INFO: terminating due to SIGCHLD

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core>
[main.c:807]: INFO: signal 15 received

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core>
[main.c:807]: INFO: signal 15 received
...
--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

Without ldap.so loaded, we can start the kamailio as a service normally with
"service kamailio start".
The platform running kamailio is: RHEL5.6_X64


Can you help us please? Because we want to start the kamailio as a service
from the boot


Thanks a lot in advanced!


Best Regards,
Laura
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From miconda at gmail.com  Tue Jun 21 11:48:46 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 11:48:46 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
Message-ID: <4E0068FE.1060904@gmail.com>

Hello,

the first error I could spot related to ldap is:

Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap 
[ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact 
LDAP server

suggesting connectivity problem to ldap server. Is it running? Can you 
connect to ldap server from a terminal?

Cheers,
Daniel


On 6/21/11 11:39 AM, laura testi wrote:
> Hi,
> we try to use LDAP module in kamailio, first we try only load the 
> ldap.so only to see if it can be loaded. When we start from the 
> command line with either "kamctl start" or directly "kamilio", the 
> kamailio is started normally with different children. When I try to 
> start as service with "service kamailio start" or 
> "/etc/init.d/kamailio start". It fails with the following error in the 
> /var/log/messages:
> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]: 
> init_tcp: using epoll_lt as the io watch method (auto detected)
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc 
> [hslot.c:53]: locks array size 512
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth 
> [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support 
> disabled
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: 
> permissions [parse_config.c:251]: file not found: 
> //etc/kamailio/permissions.allow
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: 
> permissions [permissions.c:606]: default allow file 
> (//etc/kamailio/permissions.allow) not found => empty rule set
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: 
> permissions [parse_config.c:251]: file not found: 
> //etc/kamailio/permissions.deny
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: 
> permissions [permissions.c:615]: default deny file 
> (//etc/kamailio/permissions.deny) not found => empty rule set
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap 
> [ldap_mod.c:246]: OpenLDAP - 20343
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_init.c:511]: tls: _init_tls_h:compiledwithopensslversion "OpenSSL 
> 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, 
> compression: on
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_init.c:519]: tls: init_tls_h: installed openssl library version 
> "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos 
> support: on,zlib compression: oncompiler: gcc -fPIC -DOPENSSL_PIC 
> -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H 
> -DKRB5_MIT -I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall 
> -DMD32_REG_T=int -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 
> -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 
> -mtune=generic -Wa,--noexecstack -DOPENSSL_USE_NEW_FUNCTIONS 
> -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT -DSHA1_ASM -DSHA256_ASM 
> -DSHA512_ASM -DMD5_ASM -DAES_ASM
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_init.c:373]: tls: init_tls: disabling compression...
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls 
> [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug 
> detected,kerberos support will be disabled...
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls 
> [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low 
> memory) workaround enabled (on low memory tls operations will fail 
> preemptively) with free memory thresholds 9961472 and 4980736 bytes
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has 
> been changed to 9961472
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has 
> been changed to 4980736
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:176]: TLSs<default>: tls_method=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:188]: TLSs<default>: 
> certificate='//etc/kamailio/FBCIDS01V01.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:202]: TLSs<default>: crl='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:206]: TLSs<default>: require_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:213]: TLSs<default>: cipher_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:220]: TLSs<default>: 
> private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:224]: TLSs<default>: verify_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:227]: TLSs<default>: verify_depth=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:544]: TLSs<default>: No client certificate required and 
> no checks performed
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version 
> (90802f)has the SSL_OP_TLS_BLOCK_PADDING bug, but compressionis 
> disabled so no workaround is needed
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:176]: TLSc<default>: tls_method=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:188]: TLSc<default>: certificate='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:195]: TLSc<default>: ca_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:202]: TLSc<default>: crl='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:206]: TLSc<default>: require_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:213]: TLSc<default>: cipher_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:220]: TLSc<default>: private_key='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:224]: TLSc<default>: verify_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:227]: TLSc<default>: verify_depth=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version 
> (90802f)has the SSL_OP_TLS_BLOCK_PADDING bug, but compressionis 
> disabled so no workaround is needed
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:176]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> tls_method=12
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:188]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> certificate='//etc/kamailio/FBCIDS01V01.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:195]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> ca_list='//etc/kamailio/tiglobalca.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:202]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> crl='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:206]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> require_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:213]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> cipher_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:220]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:224]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> verify_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:227]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> verify_depth=9
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:544]: TLSs<10.10.5.214:5061 <http://10.10.5.214:5061>>: 
> No client certificate required and no checks performed
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls 
> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version 
> (90802f)has the SSL_OP_TLS_BLOCK_PADDING bug, but compressionis 
> disabled so no workaround is needed
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap 
> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact 
> LDAP server
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap 
> [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> 
> [sr_module.c:832]: init_mod_child(): Error while initializing module 
> ldap (/usr/lib64/kamailio/modules_k/ldap.so)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> 
> [pt.c:337]: ERROR: fork_process(): init_child failed forprocess 1, pid 
> 8786, "udp receiver child=0 sock=127.0.0.1:5060 <http://127.0.0.1:5060>"
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core> 
> [main.c:1546]: main_loop: Cannot fork
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap 
> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact 
> LDAP server
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap 
> [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> 
> [sr_module.c:832]: init_mod_child(): Error while initializing module 
> ldap (/usr/lib64/kamailio/modules_k/ldap.so)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> 
> [pt.c:337]: ERROR: fork_process(): init_child failed forprocess 2, pid 
> 8788, "udp receiver child=1 sock=127.0.0.1:5060 <http://127.0.0.1:5060>"
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core> 
> [main.c:1546]: main_loop: Cannot fork
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> 
> [main.c:738]: child process 8786 exited normally, status=255
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> 
> [main.c:738]: child process 8788 exited normally, status=255
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> 
> [main.c:756]: INFO: terminating due to SIGCHLD
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core> 
> [main.c:807]: INFO: signal 15 received
>
> ...
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> Without ldap.so loaded, we can start the kamailio as a service 
> normally with "service kamailio start".
> The platform running kamailio is: RHEL5.6_X64
>
>
> Can you help us please? Because we want to start the kamailio as a 
> service from the boot
>
>
> Thanks a lot in advanced!
>
>
> Best Regards,
> Laura
>
>
>
> _______________________________________________
> sr-dev mailing list
> sr-dev at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From miconda at gmail.com  Tue Jun 21 11:55:24 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 11:55:24 +0200
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E005E62.3090205@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
Message-ID: <4E006A8C.50704@gmail.com>

Hello,

On 6/21/11 11:03 AM, vaad.fabi at gmail.com wrote:
>  Hello all,
>
> I want to use realtime architecture kamailio 3.1 + asterisk 1.6.2 
> described at 
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb,  
> but use it with another softswitch (it also use mysql db for 
> configuration).
> If it possible please answer for several questions:
>
> 1.    What is "#!define WITH_ASTERISK" option doing ?
>         Does it only use for:
>         - load uac module
>         - read login\passw table info
>         - some routing cases
>     or there are some else jobs ?
> Maybe there is some description for "#!define WITH_ASTERISK" ?

#!define XYZ - is a config preprocessor directive that can be used to 
set a value for a token or enable/disable parts of configuration file.

In this particular case if #!define WITH_ASTERISK is not in the config, 
all the parts between:

#!ifdef WITH_ASTERISK
...
#!endif

will be ignored by kamailio.

>
>
> 2.    What else fields of asterisk realtime table (except 
> username\sippasswd) kamailio uses for integration ?
This two are enough for authentication.

>
> 3.    From integration manual:
>         "
>         CREATE TABLE version (
>         table_name VARCHAR(32) NOT NULL,
>         table_version INT UNSIGNED DEFAULT 0 NOT NULL
>         );
>         INSERT INTO version (table_name, table_version) VALUES 
> ('sipusers','6');
>         "
> For what purposes this table used ? How to change it for using with 
> another softswitch ?
This table is required by kamailio to check the version of tables used 
internally - default one for this case would be subscriber table, but 
since it connects to asterisk database and uses instead the sipusers 
table, it has to be able to see that the version for sipusers table 
structure is 6.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From lau.testi at gmail.com  Tue Jun 21 12:00:32 2011
From: lau.testi at gmail.com (laura testi)
Date: Tue, 21 Jun 2011 12:00:32 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <4E0068FE.1060904@gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
Message-ID: <BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>

Hi Daniel,
thank you very much for your quick reply. Yes I can connect to ldap and make
search from kamailio if I start the kamailio with command line in a terminal
("# kamctl start" or "#kamailio") as root.It fail to connect to ldap only
when I start it as service from the boot or I run the command "service
kamailio start".

Best Regards,
Peng



On Tue, Jun 21, 2011 at 11:48 AM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

>  Hello,
>
> the first error I could spot related to ldap is:
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
> server
>
> suggesting connectivity problem to ldap server. Is it running? Can you
> connect to ldap server from a terminal?
>
> Cheers,
> Daniel
>
>
> On 6/21/11 11:39 AM, laura testi wrote:
>
> Hi,
> we try to use LDAP module in kamailio, first we try only load the ldap.so
> only to see if it can be loaded. When we start from the command line with
> either "kamctl start" or directly "kamilio", the kamailio is started
> normally with different children. When I try to start as service with
> "service kamailio start" or "/etc/init.d/kamailio start". It fails with the
> following error in the /var/log/messages:
>
> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]:
> init_tcp: using epoll_lt as the io watch method (auto detected)
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc
> [hslot.c:53]: locks array size 512
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth
> [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support
> disabled
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
> [parse_config.c:251]: file not found: //etc/kamailio/permissions.allow
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
> [permissions.c:606]: default allow file (//etc/kamailio/permissions.allow)
> not found => empty rule set
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
> [parse_config.c:251]: file not found: //etc/kamailio/permissions.deny
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
> [permissions.c:615]: default deny file (//etc/kamailio/permissions.deny) not
> found => empty rule set
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap
> [ldap_mod.c:246]: OpenLDAP - 20343
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_init.c:511]: tls: _init_tls_h:  compiled  with  openssl  version
> "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on,
> compression: on
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_init.c:519]: tls: init_tls_h: installed openssl library version
> "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on,
> zlib compression: on  compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB
> -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT
> -I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall -DMD32_REG_T=int -O2 -g
> -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector
> --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack
> -DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT
> -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_init.c:373]: tls: init_tls: disabling compression...
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls
> [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug detected,  kerberos
> support will be disabled...
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls
> [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory)
> workaround enabled (on low memory tls operations will fail preemptively)
> with free memory thresholds 9961472 and 4980736 bytes
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been
> changed to 9961472
>
> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been
> changed to 4980736
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:176]: TLSs<default>: tls_method=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:188]: TLSs<default>:
> certificate='//etc/kamailio/FBCIDS01V01.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:202]: TLSs<default>: crl='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:206]: TLSs<default>: require_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:213]: TLSs<default>: cipher_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:220]: TLSs<default>:
> private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:224]: TLSs<default>: verify_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:227]: TLSs<default>: verify_depth=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:544]: TLSs<default>: No client certificate required and no
> checks performed
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
> has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
> workaround is needed
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:176]: TLSc<default>: tls_method=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:188]: TLSc<default>: certificate='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:195]: TLSc<default>: ca_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:202]: TLSc<default>: crl='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:206]: TLSc<default>: require_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:213]: TLSc<default>: cipher_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:220]: TLSc<default>: private_key='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:224]: TLSc<default>: verify_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:227]: TLSc<default>: verify_depth=9
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
> has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
> workaround is needed
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:176]: TLSs<10.10.5.214:5061>: tls_method=12
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:188]: TLSs<10.10.5.214:5061>:
> certificate='//etc/kamailio/FBCIDS01V01.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:195]: TLSs<10.10.5.214:5061>:
> ca_list='//etc/kamailio/tiglobalca.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:202]: TLSs<10.10.5.214:5061>: crl='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:206]: TLSs<10.10.5.214:5061>: require_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:213]: TLSs<10.10.5.214:5061>: cipher_list='(null)'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:220]: TLSs<10.10.5.214:5061>:
> private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:224]: TLSs<10.10.5.214:5061>: verify_certificate=0
>
> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:227]: TLSs<10.10.5.214:5061>: verify_depth=9
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:544]: TLSs<10.10.5.214:5061>: No client certificate required
> and no checks performed
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
> has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
> workaround is needed
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
> server
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
> [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core>
> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap
> (/usr/lib64/kamailio/modules_k/ldap.so)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core>
> [pt.c:337]: ERROR: fork_process(): init_child failed for  process 1, pid
> 8786, "udp receiver child=0 sock=127.0.0.1:5060"
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core>
> [main.c:1546]: main_loop: Cannot fork
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
> server
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
> [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core>
> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap
> (/usr/lib64/kamailio/modules_k/ldap.so)
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core>
> [pt.c:337]: ERROR: fork_process(): init_child failed for  process 2, pid
> 8788, "udp receiver child=1 sock=127.0.0.1:5060"
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core>
> [main.c:1546]: main_loop: Cannot fork
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core>
> [main.c:738]: child process 8786 exited normally, status=255
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core>
> [main.c:738]: child process 8788 exited normally, status=255
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
> [main.c:756]: INFO: terminating due to SIGCHLD
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
>
> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core>
> [main.c:807]: INFO: signal 15 received
> ...
>
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> Without ldap.so loaded, we can start the kamailio as a service normally
> with "service kamailio start".
> The platform running kamailio is: RHEL5.6_X64
>
>
> Can you help us please? Because we want to start the kamailio as a service
> from the boot
>
>
> Thanks a lot in advanced!
>
>
> Best Regards,
> Laura
>
>
>
> _______________________________________________
> sr-dev mailing listsr-dev at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From miconda at gmail.com  Tue Jun 21 12:07:00 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 12:07:00 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
Message-ID: <4E006D44.4000802@gmail.com>

Hello,

interesting, is the ldap server running on same server? Is the 
connection via local network?

When starting as service, kamailio is probably run under user kamailio. 
Any firewall in place or system restrictions for user 'kamailio'?

You can change the user/group for kamailio service in /etc/default/kamailio

Cheers,
Daniel

On 6/21/11 12:00 PM, laura testi wrote:
> Hi Daniel,
> thank you very much for your quick reply. Yes I can connect to ldap 
> and make search from kamailio if I start the kamailio with command 
> line in a terminal ("# kamctl start" or "#kamailio") as root.It fail 
> to connect to ldap only when I start it as service from the boot or I 
> run the command "service kamailio start".
>
> Best Regards,
> Peng
>
>
>
> On Tue, Jun 21, 2011 at 11:48 AM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     the first error I could spot related to ldap is:
>
>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
>     [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't
>     contact LDAP server
>
>     suggesting connectivity problem to ldap server. Is it running? Can
>     you connect to ldap server from a terminal?
>
>     Cheers,
>     Daniel
>
>
>     On 6/21/11 11:39 AM, laura testi wrote:
>>     Hi,
>>     we try to use LDAP module in kamailio, first we try only load the
>>     ldap.so only to see if it can be loaded. When we start from the
>>     command line with either "kamctl start" or directly "kamilio",
>>     the kamailio is started normally with different children. When I
>>     try to start as service with "service kamailio start" or
>>     "/etc/init.d/kamailio start". It fails with the following error
>>     in the /var/log/messages:
>>     ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>     Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core>
>>     [tcp_main.c:4716]: init_tcp: using epoll_lt as the io watch
>>     method (auto detected)
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     usrloc [hslot.c:53]: locks array size 512
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth
>>     [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled)
>>     support disabled
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     permissions [parse_config.c:251]: file not found:
>>     //etc/kamailio/permissions.allow
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     permissions [permissions.c:606]: default allow file
>>     (//etc/kamailio/permissions.allow) not found => empty rule set
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     permissions [parse_config.c:251]: file not found:
>>     //etc/kamailio/permissions.deny
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     permissions [permissions.c:615]: default deny file
>>     (//etc/kamailio/permissions.deny) not found => empty rule set
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap
>>     [ldap_mod.c:246]: OpenLDAP - 20343
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_init.c:511]: tls: _init_tls_h:compiledwithopensslversion
>>     "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos
>>     support: on, compression: on
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_init.c:519]: tls: init_tls_h: installed openssl library
>>     version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f),
>>     kerberos support: on,zlib compression: oncompiler: gcc -fPIC
>>     -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN
>>     -DHAVE_DLFCN_H -DKRB5_MIT -I/usr/kerberos/include -DL_ENDIAN
>>     -DTERMIO -Wall -DMD32_REG_T=int -O2 -g -pipe -Wall
>>     -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector
>>     --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack
>>     -DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing
>>     -DOPENSSL_BN_ASM_MONT -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM
>>     -DMD5_ASM -DAES_ASM
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_init.c:373]: tls: init_tls: disabling compression...
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING:
>>     tls [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc
>>     bug detected,kerberos support will be disabled...
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING:
>>     tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on
>>     low memory) workaround enabled (on low memory tls operations will
>>     fail preemptively) with free memory thresholds 9961472 and
>>     4980736 bytes
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now():
>>     tls.low_mem_threshold1 has been changed to 9961472
>>
>>     Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now():
>>     tls.low_mem_threshold2 has been changed to 4980736
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially
>>     129024
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally
>>     262142
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially
>>     129024
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally
>>     262142
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:176]: TLSs<default>: tls_method=9
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:188]: TLSs<default>:
>>     certificate='//etc/kamailio/FBCIDS01V01.pem'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:195]: TLSs<default>:
>>     ca_list='//etc/kamailio/tiglobalca.pem'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:202]: TLSs<default>: crl='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:206]: TLSs<default>: require_certificate=0
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:213]: TLSs<default>: cipher_list='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:220]: TLSs<default>:
>>     private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:224]: TLSs<default>: verify_certificate=0
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:227]: TLSs<default>: verify_depth=9
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:544]: TLSs<default>: No client certificate required
>>     and no checks performed
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:597]: tls: set_ssl_options: detected openssl
>>     version (90802f)has the SSL_OP_TLS_BLOCK_PADDING bug, but
>>     compressionis disabled so no workaround is needed
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:176]: TLSc<default>: tls_method=9
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:188]: TLSc<default>: certificate='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:195]: TLSc<default>: ca_list='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:202]: TLSc<default>: crl='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:206]: TLSc<default>: require_certificate=0
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:213]: TLSc<default>: cipher_list='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:220]: TLSc<default>: private_key='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:224]: TLSc<default>: verify_certificate=0
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:227]: TLSc<default>: verify_depth=9
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:547]: TLSc<default>: Server MAY present invalid
>>     certificate
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:597]: tls: set_ssl_options: detected openssl
>>     version (90802f)has the SSL_OP_TLS_BLOCK_PADDING bug, but
>>     compressionis disabled so no workaround is needed
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:176]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: tls_method=12
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:188]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>:
>>     certificate='//etc/kamailio/FBCIDS01V01.pem'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:195]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: ca_list='//etc/kamailio/tiglobalca.pem'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:202]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: crl='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:206]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: require_certificate=0
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:213]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: cipher_list='(null)'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:220]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>:
>>     private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:224]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: verify_certificate=0
>>
>>     Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:227]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: verify_depth=9
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:544]: TLSs<10.10.5.214:5061
>>     <http://10.10.5.214:5061>>: No client certificate required and no
>>     checks performed
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>>     [tls_domain.c:597]: tls: set_ssl_options: detected openssl
>>     version (90802f)has the SSL_OP_TLS_BLOCK_PADDING bug, but
>>     compressionis disabled so no workaround is needed
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
>>     [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't
>>     contact LDAP server
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
>>     [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR:
>>     <core> [sr_module.c:832]: init_mod_child(): Error while
>>     initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so)
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR:
>>     <core> [pt.c:337]: ERROR: fork_process(): init_child failed
>>     forprocess 1, pid 8786, "udp receiver child=0 sock=127.0.0.1:5060
>>     <http://127.0.0.1:5060>"
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core>
>>     [main.c:1546]: main_loop: Cannot fork
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
>>     [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't
>>     contact LDAP server
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
>>     [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR:
>>     <core> [sr_module.c:832]: init_mod_child(): Error while
>>     initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so)
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR:
>>     <core> [pt.c:337]: ERROR: fork_process(): init_child failed
>>     forprocess 2, pid 8788, "udp receiver child=1 sock=127.0.0.1:5060
>>     <http://127.0.0.1:5060>"
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core>
>>     [main.c:1546]: main_loop: Cannot fork
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT:
>>     <core> [main.c:738]: child process 8786 exited normally, status=255
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT:
>>     <core> [main.c:738]: child process 8788 exited normally, status=255
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO:
>>     <core> [main.c:756]: INFO: terminating due to SIGCHLD
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO:
>>     <core> [main.c:807]: INFO: signal 15 received
>>
>>     ...
>>     --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>     Without ldap.so loaded, we can start the kamailio as a service
>>     normally with "service kamailio start".
>>     The platform running kamailio is: RHEL5.6_X64
>>
>>
>>     Can you help us please? Because we want to start the kamailio as
>>     a service from the boot
>>
>>
>>     Thanks a lot in advanced!
>>
>>
>>     Best Regards,
>>     Laura
>>
>>
>>
>>     _______________________________________________
>>     sr-dev mailing list
>>     sr-dev at lists.sip-router.org  <mailto:sr-dev at lists.sip-router.org>
>>     http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>
>     -- 
>     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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From vaad.fabi at gmail.com  Tue Jun 21 12:17:46 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Tue, 21 Jun 2011 13:17:46 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E006A8C.50704@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com>
Message-ID: <4E006FCA.9010904@gmail.com>

  On 06/21/2011 12:55 PM, Daniel-Constantin Mierla wrote:
>>
>> 3.    From integration manual:
>>         "
>>         CREATE TABLE version (
>>         table_name VARCHAR(32) NOT NULL,
>>         table_version INT UNSIGNED DEFAULT 0 NOT NULL
>>         );
>>         INSERT INTO version (table_name, table_version) VALUES 
>> ('sipusers','6');
>>         "
>> For what purposes this table used ? How to change it for using with 
>> another softswitch ?
> This table is required by kamailio to check the version of tables used 
> internally - default one for this case would be subscriber table, but 
> since it connects to asterisk database and uses instead the sipusers 
> table, it has to be able to see that the version for sipusers table 
> structure is 6.
>
> Cheers,
> Daniel

Daniel, thanx for your help and wonderful software!



I want to use different table structure (non asterisk) for integration, 
and the questions are:
  - what sipusers table fields it uses (only username\sippasswd or some 
else)
  - where kamailio save registration information (in sipusers or in 
another own table)

-- 
vf



From lau.testi at gmail.com  Tue Jun 21 12:33:08 2011
From: lau.testi at gmail.com (laura testi)
Date: Tue, 21 Jun 2011 12:33:08 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <4E006D44.4000802@gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
	<4E006D44.4000802@gmail.com>
Message-ID: <BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>

The LDAP server is running on another machine in the same LAN without any
firewall.

As by your suggestion, I change the user and group from kamailio to root in
the /etc/default/kamilio file. Now it works.

Why I can not run the kamailio as the user kamailio as the default
installation set? (I install the kamilio from rpm and the user and group of
kamailio are already created by the installation)

wuthout ldap.so, I can run the kamilio from the user kamailio normally when
I run "service kamailio start" from root.





On Tue, Jun 21, 2011 at 12:07 PM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

>  Hello,
>
> interesting, is the ldap server running on same server? Is the connection
> via local network?
>
> When starting as service, kamailio is probably run under user kamailio. Any
> firewall in place or system restrictions for user 'kamailio'?
>
> You can change the user/group for kamailio service in /etc/default/kamailio
>
> Cheers,
> Daniel
>
>
> On 6/21/11 12:00 PM, laura testi wrote:
>
> Hi Daniel,
> thank you very much for your quick reply. Yes I can connect to ldap and
> make search from kamailio if I start the kamailio with command line in a
> terminal ("# kamctl start" or "#kamailio") as root.It fail to connect to
> ldap only when I start it as service from the boot or I run the command
> "service kamailio start".
>
> Best Regards,
> Peng
>
>
>
> On Tue, Jun 21, 2011 at 11:48 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> the first error I could spot related to ldap is:
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
>> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
>> server
>>
>>  suggesting connectivity problem to ldap server. Is it running? Can you
>> connect to ldap server from a terminal?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 6/21/11 11:39 AM, laura testi wrote:
>>
>>  Hi,
>> we try to use LDAP module in kamailio, first we try only load the ldap.so
>> only to see if it can be loaded. When we start from the command line with
>> either "kamctl start" or directly "kamilio", the kamailio is started
>> normally with different children. When I try to start as service with
>> "service kamailio start" or "/etc/init.d/kamailio start". It fails with the
>> following error in the /var/log/messages:
>>
>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>
>> Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]:
>> init_tcp: using epoll_lt as the io watch method (auto detected)
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc
>> [hslot.c:53]: locks array size 512
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth
>> [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support
>> disabled
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
>> [parse_config.c:251]: file not found: //etc/kamailio/permissions.allow
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
>> [permissions.c:606]: default allow file (//etc/kamailio/permissions.allow)
>> not found => empty rule set
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
>> [parse_config.c:251]: file not found: //etc/kamailio/permissions.deny
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions
>> [permissions.c:615]: default deny file (//etc/kamailio/permissions.deny) not
>> found => empty rule set
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap
>> [ldap_mod.c:246]: OpenLDAP - 20343
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_init.c:511]: tls: _init_tls_h:  compiled  with  openssl  version
>> "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on,
>> compression: on
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_init.c:519]: tls: init_tls_h: installed openssl library version
>> "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on,
>> zlib compression: on  compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB
>> -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT
>> -I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall -DMD32_REG_T=int -O2 -g
>> -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector
>> --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack
>> -DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT
>> -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_init.c:373]: tls: init_tls: disabling compression...
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls
>> [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug detected,
>> kerberos support will be disabled...
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls
>> [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory)
>> workaround enabled (on low memory tls operations will fail preemptively)
>> with free memory thresholds 9961472 and 4980736 bytes
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been
>> changed to 9961472
>>
>> Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been
>> changed to 4980736
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:176]: TLSs<default>: tls_method=9
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:188]: TLSs<default>:
>> certificate='//etc/kamailio/FBCIDS01V01.pem'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:202]: TLSs<default>: crl='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:206]: TLSs<default>: require_certificate=0
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:213]: TLSs<default>: cipher_list='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:220]: TLSs<default>:
>> private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:224]: TLSs<default>: verify_certificate=0
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:227]: TLSs<default>: verify_depth=9
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:544]: TLSs<default>: No client certificate required and no
>> checks performed
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
>> has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
>> workaround is needed
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:176]: TLSc<default>: tls_method=9
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:188]: TLSc<default>: certificate='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:195]: TLSc<default>: ca_list='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:202]: TLSc<default>: crl='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:206]: TLSc<default>: require_certificate=0
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:213]: TLSc<default>: cipher_list='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:220]: TLSc<default>: private_key='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:224]: TLSc<default>: verify_certificate=0
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:227]: TLSc<default>: verify_depth=9
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
>> has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
>> workaround is needed
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:176]: TLSs<10.10.5.214:5061>: tls_method=12
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:188]: TLSs<10.10.5.214:5061>:
>> certificate='//etc/kamailio/FBCIDS01V01.pem'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:195]: TLSs<10.10.5.214:5061>:
>> ca_list='//etc/kamailio/tiglobalca.pem'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:202]: TLSs<10.10.5.214:5061>: crl='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:206]: TLSs<10.10.5.214:5061>: require_certificate=0
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:213]: TLSs<10.10.5.214:5061>: cipher_list='(null)'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:220]: TLSs<10.10.5.214:5061>:
>> private_key='//etc/kamailio/FBCIDS01V01KEY.pem'
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:224]: TLSs<10.10.5.214:5061>: verify_certificate=0
>>
>> Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:227]: TLSs<10.10.5.214:5061>: verify_depth=9
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:544]: TLSs<10.10.5.214:5061>: No client certificate
>> required and no checks performed
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls
>> [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f)
>> has the SSL_OP_TLS_BLOCK_PADDING bug, but compression  is disabled so no
>> workaround is needed
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
>> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
>> server
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap
>> [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core>
>> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap
>> (/usr/lib64/kamailio/modules_k/ldap.so)
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core>
>> [pt.c:337]: ERROR: fork_process(): init_child failed for  process 1, pid
>> 8786, "udp receiver child=0 sock=127.0.0.1:5060"
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core>
>> [main.c:1546]: main_loop: Cannot fork
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
>> [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP
>> server
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap
>> [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s)
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core>
>> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap
>> (/usr/lib64/kamailio/modules_k/ldap.so)
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core>
>> [pt.c:337]: ERROR: fork_process(): init_child failed for  process 2, pid
>> 8788, "udp receiver child=1 sock=127.0.0.1:5060"
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core>
>> [main.c:1546]: main_loop: Cannot fork
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core>
>> [main.c:738]: child process 8786 exited normally, status=255
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core>
>> [main.c:738]: child process 8788 exited normally, status=255
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core>
>> [main.c:756]: INFO: terminating due to SIGCHLD
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>>
>> Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core>
>> [main.c:807]: INFO: signal 15 received
>> ...
>>
>> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>>
>> Without ldap.so loaded, we can start the kamailio as a service normally
>> with "service kamailio start".
>> The platform running kamailio is: RHEL5.6_X64
>>
>>
>> Can you help us please? Because we want to start the kamailio as a service
>> from the boot
>>
>>
>> Thanks a lot in advanced!
>>
>>
>> Best Regards,
>> Laura
>>
>>
>>
>>  _______________________________________________
>> sr-dev mailing listsr-dev at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>>
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From klaus.mailinglists at pernau.at  Tue Jun 21 13:50:05 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 21 Jun 2011 13:50:05 +0200
Subject: [SR-Users] an IDEA
In-Reply-To: <BANLkTiktPJ4Oyzb6isep8ZSL03-Co3BHaw@mail.gmail.com>
References: <BANLkTiktPJ4Oyzb6isep8ZSL03-Co3BHaw@mail.gmail.com>
Message-ID: <4E00856D.7020504@pernau.at>

You may wanto to look at Asterisk and its SSL7 stacks (chan_ss7,
libss7+dahdi)

regards
Klaus

Am 18.06.2011 15:25, schrieb afshin afzali:
> Hi Guys,
> 
> I need to integrate a SIP processing environment to H.248 and SS7/ ISUP
> links.
> For those communication links, OpenSS7 and Erlang/OTP MEGACO are good
> solutions. My idea is to write applications over OTP and SIP-Router to
> be able to bridge non-SIP protocols to SIP!
> On OTP the way is clear, but on SIP-Router I need to be able to generate
> / terminate SIP transactions. I'll appreciate every comment.
> 
> BEST,
> -- afshin 
> 
> 
> 
> _______________________________________________
> 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 Jun 21 14:30:52 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 14:30:52 +0200
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E006FCA.9010904@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
Message-ID: <4E008EFC.3020001@gmail.com>



On 6/21/11 12:17 PM, vaad.fabi at gmail.com wrote:
>  On 06/21/2011 12:55 PM, Daniel-Constantin Mierla wrote:
>>>
>>> 3.    From integration manual:
>>>         "
>>>         CREATE TABLE version (
>>>         table_name VARCHAR(32) NOT NULL,
>>>         table_version INT UNSIGNED DEFAULT 0 NOT NULL
>>>         );
>>>         INSERT INTO version (table_name, table_version) VALUES 
>>> ('sipusers','6');
>>>         "
>>> For what purposes this table used ? How to change it for using with 
>>> another softswitch ?
>> This table is required by kamailio to check the version of tables 
>> used internally - default one for this case would be subscriber 
>> table, but since it connects to asterisk database and uses instead 
>> the sipusers table, it has to be able to see that the version for 
>> sipusers table structure is 6.
>>
>> Cheers,
>> Daniel
>
> Daniel, thanx for your help and wonderful software!
>
>
>
> I want to use different table structure (non asterisk) for 
> integration, and the questions are:
>  - what sipusers table fields it uses (only username\sippasswd or some 
> else)
only these two,

>  - where kamailio save registration information (in sipusers or in 
> another own table)
it keeps in its own 'location' table located in openser database.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Tue Jun 21 14:33:32 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Jun 2011 14:33:32 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
	<4E006D44.4000802@gmail.com>
	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>
Message-ID: <4E008F9C.2000101@gmail.com>



On 6/21/11 12:33 PM, laura testi wrote:
> The LDAP server is running on another machine in the same LAN without 
> any firewall.
>
> As by your suggestion, I change the user and group from kamailio to 
> root in the /etc/default/kamilio file. Now it works.
>
> Why I can not run the kamailio as the user kamailio as the default 
> installation set? (I install the kamilio from rpm and the user and 
> group of kamailio are already created by the installation)
>
> wuthout ldap.so, I can run the kamilio from the user kamailio normally 
> when I run "service kamailio start" from root.
Is the ldap module configuration file readable by user kamailio?

Cheers,
Daniel

>
>
> On Tue, Jun 21, 2011 at 12:07 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     interesting, is the ldap server running on same server? Is the
>     connection via local network?
>
>     When starting as service, kamailio is probably run under user
>     kamailio. Any firewall in place or system restrictions for user
>     'kamailio'?
>
>     You can change the user/group for kamailio service in
>     /etc/default/kamailio
>
>     Cheers,
>     Daniel
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From lau.testi at gmail.com  Tue Jun 21 16:08:40 2011
From: lau.testi at gmail.com (laura testi)
Date: Tue, 21 Jun 2011 16:08:40 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <4E008F9C.2000101@gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
	<4E006D44.4000802@gmail.com>
	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>
	<4E008F9C.2000101@gmail.com>
Message-ID: <BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>

Yes, it can be read by all user like other configuration file:
-rw-r--r--   1 root root   566 Jun 20 18:56 ldap.cfg

On Tue, Jun 21, 2011 at 2:33 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>
>
> On 6/21/11 12:33 PM, laura testi wrote:
>
> The LDAP server is running on another machine in the same LAN without any
> firewall.
>
> As by your suggestion, I change the user and group from kamailio to root in
> the /etc/default/kamilio file. Now it works.
>
> Why I can not run the kamailio as the user kamailio as the default
> installation set? (I install the kamilio from rpm and the user and group of
> kamailio are already created by the installation)
>
> wuthout ldap.so, I can run the kamilio from the user kamailio normally when
> I run "service kamailio start" from root.
>
> Is the ldap module configuration file readable by user kamailio?
>
> Cheers,
> Daniel
>
>
>
>
> On Tue, Jun 21, 2011 at 12:07 PM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>>  Hello,
>>
>> interesting, is the ldap server running on same server? Is the connection
>> via local network?
>>
>> When starting as service, kamailio is probably run under user kamailio.
>> Any firewall in place or system restrictions for user 'kamailio'?
>>
>> You can change the user/group for kamailio service in
>> /etc/default/kamailio
>>
>> Cheers,
>> Daniel
>>
>>
> --
> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From andrew.pogrebennyk at portaone.com  Tue Jun 21 16:23:43 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Tue, 21 Jun 2011 17:23:43 +0300
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>	<4E0068FE.1060904@gmail.com>	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>	<4E006D44.4000802@gmail.com>	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>	<4E008F9C.2000101@gmail.com>
	<BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>
Message-ID: <4E00A96F.4000903@portaone.com>

On 21.06.2011 17:08, laura testi wrote:
> Yes, it can be read by all user like other configuration file:
> -rw-r--r--   1 root root   566 Jun 20 18:56 ldap.cfg

Just in case, have you set perhaps any of the environment variables 
manually for working with ldap? 'service' command is a wrapper that 
calls env -i (start with an empty environment), maybe this is the reason 
it works differently for the root and kamailio users.

-- 
Sincerely,
Andrew Pogrebennyk


From lau.testi at gmail.com  Tue Jun 21 16:30:42 2011
From: lau.testi at gmail.com (laura testi)
Date: Tue, 21 Jun 2011 16:30:42 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <4E00A96F.4000903@portaone.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
	<4E006D44.4000802@gmail.com>
	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>
	<4E008F9C.2000101@gmail.com>
	<BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>
	<4E00A96F.4000903@portaone.com>
Message-ID: <BANLkTimLVQuc8nkmW_oDEs6QNZpgWuz1KQ@mail.gmail.com>

the env retrun by root user is:
[root at kamailio3p1 kamailio]# env
HOSTNAME=kamailio3p1.emslabw3.local
TERM=xterm
SHELL=/bin/bash
HISTSIZE=1000
SSH_CLIENT=10.10.2.28 49471 22
SSH_TTY=/dev/pts/5
USER=root
LS_COLORS=no=00:fi=00:di=00;34:ln=00;36:pi=40;33:so=00;35:bd=40;33;01:cd=40;33;01:or=01;05;37;41:mi=01;05;37;41:ex=00;32:*.cmd=00;32:*.exe=00;32:*.com=00;32:*.btm=00;32:*.bat=00;32:*.sh=00;32:*.csh=00;32:*.tar=00;31:*.tgz=00;31:*.arj=00;31:*.taz=00;31:*.lzh=00;31:*.zip=00;31:*.z=00;31:*.Z=00;31:*.gz=00;31:*.bz2=00;31:*.bz=00;31:*.tz=00;31:*.rpm=00;31:*.cpio=00;31:*.jpg=00;35:*.gif=00;35:*.bmp=00;35:*.xbm=00;35:*.xpm=00;35:*.png=00;35:*.tif=00;35:
MAIL=/var/spool/mail/root
PATH=/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin
INPUTRC=/etc/inputrc
PWD=/etc/kamailio
LANG=en_US.UTF-8
SSH_ASKPASS=/usr/libexec/openssh/gnome-ssh-askpass
SHLVL=1
HOME=/root
LOGNAME=root
CVS_RSH=ssh
SSH_CONNECTION=10.10.2.28 49471 10.10.5.214 22
LESSOPEN=|/usr/bin/lesspipe.sh %s
DISPLAY=localhost:15.0
G_BROKEN_FILENAMES=1
OLDPWD=/root
_=/bin/env


while env -i is empty.

Should I need to set a some special environment variable to load ldap.so in
the /etc/init.d/kamailio?

Thanks,
Laura

On Tue, Jun 21, 2011 at 4:23 PM, Andrew Pogrebennyk <
andrew.pogrebennyk at portaone.com> wrote:

> On 21.06.2011 17:08, laura testi wrote:
>
>> Yes, it can be read by all user like other configuration file:
>> -rw-r--r--   1 root root   566 Jun 20 18:56 ldap.cfg
>>
>
> Just in case, have you set perhaps any of the environment variables
> manually for working with ldap? 'service' command is a wrapper that calls
> env -i (start with an empty environment), maybe this is the reason it works
> differently for the root and kamailio users.
>
> --
> Sincerely,
> Andrew Pogrebennyk
>
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From a.afzali2003 at gmail.com  Tue Jun 21 17:16:40 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 21 Jun 2011 19:46:40 +0430
Subject: [SR-Users] an IDEA
In-Reply-To: <4E00856D.7020504@pernau.at>
References: <BANLkTiktPJ4Oyzb6isep8ZSL03-Co3BHaw@mail.gmail.com>
	<4E00856D.7020504@pernau.at>
Message-ID: <BANLkTi=LmALG9tnTE9srJ4i5GC1sqv7fNw@mail.gmail.com>

Thanks to all.
-- afshin

On Tue, Jun 21, 2011 at 4:20 PM, Klaus Darilion <
klaus.mailinglists at pernau.at> wrote:

> You may wanto to look at Asterisk and its SSL7 stacks (chan_ss7,
> libss7+dahdi)
>
> regards
> Klaus
>
> Am 18.06.2011 15:25, schrieb afshin afzali:
> > Hi Guys,
> >
> > I need to integrate a SIP processing environment to H.248 and SS7/ ISUP
> > links.
> > For those communication links, OpenSS7 and Erlang/OTP MEGACO are good
> > solutions. My idea is to write applications over OTP and SIP-Router to
> > be able to bridge non-SIP protocols to SIP!
> > On OTP the way is clear, but on SIP-Router I need to be able to generate
> > / terminate SIP transactions. I'll appreciate every comment.
> >
> > BEST,
> > -- afshin
> >
> >
> >
> > _______________________________________________
> > 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 andrew.pogrebennyk at portaone.com  Tue Jun 21 17:26:35 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Tue, 21 Jun 2011 18:26:35 +0300
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <BANLkTimLVQuc8nkmW_oDEs6QNZpgWuz1KQ@mail.gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>	<4E0068FE.1060904@gmail.com>	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>	<4E006D44.4000802@gmail.com>	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>	<4E008F9C.2000101@gmail.com>	<BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>	<4E00A96F.4000903@portaone.com>
	<BANLkTimLVQuc8nkmW_oDEs6QNZpgWuz1KQ@mail.gmail.com>
Message-ID: <4E00B82B.5070102@portaone.com>

On 21.06.2011 17:30, laura testi wrote:
> Should I need to set a some special environment variable to load ldap.so
> in the /etc/init.d/kamailio?

The env output for root does not confirm my theory :) I'm not sure what 
is preventing kamailio from communication with LDAP, could it be that 
SELinux is enabled and denying it in some way?

-- 
Sincerely,
Andrew Pogrebennyk


From henning.westerholt at 1und1.de  Tue Jun 21 17:30:14 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Tue, 21 Jun 2011 17:30:14 +0200
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
In-Reply-To: <8869B5C4-D4D4-4B06-9015-7E753BC68E6C@321communications.com>
References: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>
	<201106201111.11708.henning.westerholt@1und1.de>
	<8869B5C4-D4D4-4B06-9015-7E753BC68E6C@321communications.com>
Message-ID: <201106211730.14087.henning.westerholt@1und1.de>

On Monday 20 June 2011, Omar wrote:
> Everything compiles fine until the showed error the were errors about the
> libconfuse which I fixed stating the path to the library, yes lib confuse
> is installed I am using the same one I installed for the kamailio 1.4.3.

Hi Omar,

ok, strange. If it works for the 1.4.3 it should also work for the new 
version, at least i'm not aware of any changes in this regards. Can you please 
provide the output of a "make clean; make quiet=0" command in the 
modules/carrierroute directory?

Best regards,

Henning


From jh at tutpro.com  Tue Jun 21 17:31:26 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Tue, 21 Jun 2011 18:31:26 +0300
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <4E00B82B.5070102@portaone.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
	<4E006D44.4000802@gmail.com>
	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>
	<4E008F9C.2000101@gmail.com>
	<BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>
	<4E00A96F.4000903@portaone.com>
	<BANLkTimLVQuc8nkmW_oDEs6QNZpgWuz1KQ@mail.gmail.com>
	<4E00B82B.5070102@portaone.com>
Message-ID: <19968.47438.476068.6395@sip.test.fi>

could you please decide a single mailing list for this discussion.

-- juha


From omar at 321communications.com  Wed Jun 22 02:41:00 2011
From: omar at 321communications.com (Omar)
Date: Tue, 21 Jun 2011 20:41:00 -0400
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
In-Reply-To: <201106211730.14087.henning.westerholt@1und1.de>
References: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>
	<201106201111.11708.henning.westerholt@1und1.de>
	<8869B5C4-D4D4-4B06-9015-7E753BC68E6C@321communications.com>
	<201106211730.14087.henning.westerholt@1und1.de>
Message-ID: <9FB43C00-92BF-445D-970F-CC7245D4E501@321communications.com>

Here we go

bash-3.00# make clean
bash-3.00# make quiet=0
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c prime_hash.c -o prime_hash.o -MMD -MP
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c cr_data.c -o cr_data.o -MMD -MP
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c carrierroute.c -o carrierroute.o -MMD -MP
carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a function)
carrierroute.c:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cmds'
carrierroute.c:132: error: variable 'exports' has initializer but incomplete type
carrierroute.c:133: warning: excess elements in struct initializer
carrierroute.c:133: warning: (near initialization for 'exports')
carrierroute.c:134: warning: excess elements in struct initializer
carrierroute.c:134: warning: (near initialization for 'exports')
carrierroute.c:135: error: 'cmds' undeclared here (not in a function)
carrierroute.c:135: warning: excess elements in struct initializer
carrierroute.c:135: warning: (near initialization for 'exports')
carrierroute.c:136: warning: excess elements in struct initializer
carrierroute.c:136: warning: (near initialization for 'exports')
carrierroute.c:137: warning: excess elements in struct initializer
carrierroute.c:137: warning: (near initialization for 'exports')
carrierroute.c:138: warning: excess elements in struct initializer
carrierroute.c:138: warning: (near initialization for 'exports')
carrierroute.c:139: warning: excess elements in struct initializer
carrierroute.c:139: warning: (near initialization for 'exports')
carrierroute.c:140: warning: excess elements in struct initializer
carrierroute.c:140: warning: (near initialization for 'exports')
carrierroute.c:141: warning: excess elements in struct initializer
carrierroute.c:141: warning: (near initialization for 'exports')
carrierroute.c:142: warning: excess elements in struct initializer
carrierroute.c:142: warning: (near initialization for 'exports')
carrierroute.c:143: warning: excess elements in struct initializer
carrierroute.c:143: warning: (near initialization for 'exports')
carrierroute.c:145: warning: excess elements in struct initializer
carrierroute.c:145: warning: (near initialization for 'exports')
carrierroute.c: In function 'mod_init':
carrierroute.c:161: error: invalid use of undefined type 'struct module_exports'
make: *** [carrierroute.o] Error 1
bash-3.00# 

On Jun 21, 2011, at 11:30 AM, Henning Westerholt wrote:

> On Monday 20 June 2011, Omar wrote:
>> Everything compiles fine until the showed error the were errors about the
>> libconfuse which I fixed stating the path to the library, yes lib confuse
>> is installed I am using the same one I installed for the kamailio 1.4.3.
> 
> Hi Omar,
> 
> ok, strange. If it works for the 1.4.3 it should also work for the new 
> version, at least i'm not aware of any changes in this regards. Can you please 
> provide the output of a "make clean; make quiet=0" command in the 
> modules/carrierroute directory?
> 
> Best regards,
> 
> Henning
> 



From marius.zbihlei at 1and1.ro  Wed Jun 22 09:48:08 2011
From: marius.zbihlei at 1and1.ro (marius zbihlei)
Date: Wed, 22 Jun 2011 10:48:08 +0300
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
In-Reply-To: <9FB43C00-92BF-445D-970F-CC7245D4E501@321communications.com>
References: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>	<201106201111.11708.henning.westerholt@1und1.de>	<8869B5C4-D4D4-4B06-9015-7E753BC68E6C@321communications.com>	<201106211730.14087.henning.westerholt@1und1.de>
	<9FB43C00-92BF-445D-970F-CC7245D4E501@321communications.com>
Message-ID: <4E019E38.1050407@1and1.ro>

On 06/22/2011 03:41 AM, Omar wrote:
> Here we go
>
>    

Hello Omar,

You are missing a -DOPENSER_MOD_INTERFACE in the gcc command line. This 
is set in modules/carrierroute/Makefile  like below

...
DEFS+=$(CONFUSEDEFS)
LIBS=$(CONFUSELIBS)

DEFS+=-DOPENSER_MOD_INTERFACE

SERLIBPATH=../../lib
SER_LIBS+=$(SERLIBPATH)/trie/trie $(SERLIBPATH)/srdb1/srdb1 
$(SERLIBPATH)/kmi/kmi $(SERLIBPATH)/kcore/kcore

include ../../Makefile.modules
....

(This is just under the libconfuse find).
Can you check that this Makefile is unchanged(or at least you have this 
line). Also try to run make QUIET=0 from the modules/carrierroute 
directory and see what -D flags are passed to the compiler

Thanks
Marius
> bash-3.00# make clean
> bash-3.00# make quiet=0
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c prime_hash.c -o prime_hash.o -MMD -MP
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c cr_data.c -o cr_data.o -MMD -MP
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c carrierroute.c -o carrierroute.o -MMD -MP
> carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a function)
> carrierroute.c:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cmds'
> carrierroute.c:132: error: variable 'exports' has initializer but incomplete type
> carrierroute.c:133: warning: excess elements in struct initializer
> carrierroute.c:133: warning: (near initialization for 'exports')
> carrierroute.c:134: warning: excess elements in struct initializer
> carrierroute.c:134: warning: (near initialization for 'exports')
> carrierroute.c:135: error: 'cmds' undeclared here (not in a function)
> carrierroute.c:135: warning: excess elements in struct initializer
> carrierroute.c:135: warning: (near initialization for 'exports')
> carrierroute.c:136: warning: excess elements in struct initializer
> carrierroute.c:136: warning: (near initialization for 'exports')
> carrierroute.c:137: warning: excess elements in struct initializer
> carrierroute.c:137: warning: (near initialization for 'exports')
> carrierroute.c:138: warning: excess elements in struct initializer
> carrierroute.c:138: warning: (near initialization for 'exports')
> carrierroute.c:139: warning: excess elements in struct initializer
> carrierroute.c:139: warning: (near initialization for 'exports')
> carrierroute.c:140: warning: excess elements in struct initializer
> carrierroute.c:140: warning: (near initialization for 'exports')
> carrierroute.c:141: warning: excess elements in struct initializer
> carrierroute.c:141: warning: (near initialization for 'exports')
> carrierroute.c:142: warning: excess elements in struct initializer
> carrierroute.c:142: warning: (near initialization for 'exports')
> carrierroute.c:143: warning: excess elements in struct initializer
> carrierroute.c:143: warning: (near initialization for 'exports')
> carrierroute.c:145: warning: excess elements in struct initializer
> carrierroute.c:145: warning: (near initialization for 'exports')
> carrierroute.c: In function 'mod_init':
> carrierroute.c:161: error: invalid use of undefined type 'struct module_exports'
> make: *** [carrierroute.o] Error 1
> bash-3.00#
>
> On Jun 21, 2011, at 11:30 AM, Henning Westerholt wrote:
>
>    
>> On Monday 20 June 2011, Omar wrote:
>>      
>>> Everything compiles fine until the showed error the were errors about the
>>> libconfuse which I fixed stating the path to the library, yes lib confuse
>>> is installed I am using the same one I installed for the kamailio 1.4.3.
>>>        
>> Hi Omar,
>>
>> ok, strange. If it works for the 1.4.3 it should also work for the new
>> version, at least i'm not aware of any changes in this regards. Can you please
>> provide the output of a "make clean; make quiet=0" command in the
>> modules/carrierroute directory?
>>
>> Best regards,
>>
>> Henning
>>
>>      
>
> _______________________________________________
> 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 gminghon at hotmail.com  Wed Jun 22 11:34:53 2011
From: gminghon at hotmail.com (=?utf-8?B?4oCg4K6Qw7vil4/Rh8aD0YbOr9GI4oCgIEdvaA==?=)
Date: Wed, 22 Jun 2011 17:34:53 +0800
Subject: [SR-Users] audio issue in same nat.
In-Reply-To: <SNT101-W431DF375A5B660E47A0CDFCC500@phx.gbl>
References: <SNT101-W431DF375A5B660E47A0CDFCC500@phx.gbl>
Message-ID: <SNT101-W327D84CB88BE92529B0309CC500@phx.gbl>






Hi List,
below is my setup..
rtpproxy and kamailio in one PC with 2 nic. (ppp0 with public IP[60.49.119.XX] and eth1 with private IP[192.168.2.3])and asterisk is on another PC with private IP[192.168.2.23]
i use realtime integration for kamailio and asterisk.http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
i have two yealink hardphone ext 101(ip 192.168.1.200) and 102(ip 192.168.1.132) and a softphone ext 103 registered successful.both hardphone are behind same nat (175.136.221.XX)and softphone ext 103(ip 10.129.138.225) behind nat also (113.210.97.XX)
ul show:database engine 'MYSQL' loadedControl engine 'FIFO' loadedentering fifo_cmd ul_dumpDomain:: location table=512 records=3 max_slot=1        AOR:: 102                Contact:: sip:102 at 175.136.221.XX:5062 Q=                        Expires:: 3109                        Callid:: 2043273564 at 175.136.221.241                        Cseq:: 4                        User-agent:: T22 7.3.0.50                        Received:: sip:175.136.221.241:1039                        State:: CS_SYNC                        Flags:: 0                        Cflag:: 192                        Socket:: udp:60.49.119.69:5060                        Methods:: 16383        AOR:: 103                Contact:: sip:103 at 113.210.97.XX:58776;transport=UDP;ob Q=                        Expires:: 294                        Callid:: oa8Pqx3mR.SVnzAVEYHTwVKZE8CbpY9l                        Cseq:: 27626                        User-agent:: v1.0.0/iPhone                        State:: CS_NEW                        Flags:: 0                        Cflag:: 0                        Socket:: udp:60.49.119.69:5060                        Methods:: 8143        AOR:: 101                Contact:: sip:101 at 175.136.221.XX:5062 Q=                        Expires:: 1738                        Callid:: 451417581 at 175.136.221.241                        Cseq:: 2                        User-agent:: T20 9.41.0.80                        State:: CS_SYNC                        Flags:: 0                        Cflag:: 0                        Socket:: udp:60.49.119.69:5060                        Methods:: 16383FIFO command was::ul_dump:openser_receiver_17783
103 try to call 102 and 101 work fine. 101 and 102 try call 103 also fine.when 101 call 102 it work fine but when 102 call 101 there is no audio for both side.102 call 101 wireshark capture on 102 sidekeep send rtp but no receive.192.168.1.132 -> 60.49.119.XX RTP
when capture on 101 side.keep send rtp but no receive.192.168.1.200 -> 60.49.119.XX RTP
and also when 101 try to call into voicemail there is no audioit keep send rtp packet but to192.168.1.200 -> 192.168.2.23 RTP
in kamailio.cfg#!WITH_NATlisten=60.49.119.XXlisten=192.168.2.3
# uncomment next line to do SIP NAT pinging                        setbflag(FLB_NATSIPPING);
nat_uac_test("19")rtpproxy -l 60.49.119.XX -s udp:127.0.0.1 is running

anyone can help me? how can i fix this?thanks in adv.

Regards, 
minghon 		 	   		  
 		 	   		  
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From miconda at gmail.com  Wed Jun 22 11:50:32 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 22 Jun 2011 11:50:32 +0200
Subject: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service
 when load the ldap.so module
In-Reply-To: <BANLkTi=XE1D66NKKfQeSk+tn8p5ZXf=zyg@mail.gmail.com>
References: <BANLkTikpBmOAXYzGiuekXAW+4-qtiz=jew@mail.gmail.com>
	<4E0068FE.1060904@gmail.com>
	<BANLkTim_FNzcaXNPxR2YNWBd9R0Bw6JF+A@mail.gmail.com>
	<4E006D44.4000802@gmail.com>
	<BANLkTin9gT5f5mT8bm3nV6H12oyirWHGWw@mail.gmail.com>
	<4E008F9C.2000101@gmail.com>
	<BANLkTimW6=mb7P8ZzLaP4jH45SJTAcwg5Q@mail.gmail.com>
	<4E00A96F.4000903@portaone.com>
	<BANLkTimLVQuc8nkmW_oDEs6QNZpgWuz1KQ@mail.gmail.com>
	<4E00B82B.5070102@portaone.com>
	<BANLkTi=6EKf2tvoUibQ_yPBh4j1+pZ40Sg@mail.gmail.com>
	<BANLkTi=XE1D66NKKfQeSk+tn8p5ZXf=zyg@mail.gmail.com>
Message-ID: <4E01BAE8.6080907@gmail.com>

Thanks for reporting back, taking the chance to forward this email from 
sr-dev to sr-users since the thread was cross-posted to some point, for 
the benefit of future reads of the archive to show the solution of this 
case.

Cheers,
Daniel

On 6/21/11 7:19 PM, laura testi wrote:
> Found the problem after some investigation based on your suggestions. 
> In the openldap configuration file /etc/openldap/ldap.conf, the entry 
> TLS_CACERT is pointed to a certificate file in a directory which is 
> accessible only by root user. After I move the file to a directory 
> which all user can access it, now it works. I can start the kamailio 
> as a service with "service kamailio start" with the running user and 
> group is kamailio.
>
> Thank you all for your helps and suggestions!
>
> Best Regards,
> Laura
>
> On Tue, Jun 21, 2011 at 5:44 PM, laura testi <lau.testi at gmail.com 
> <mailto:lau.testi at gmail.com>> wrote:
>
>     The SELinux is disabled.
>
>
>
>
>     On Tue, Jun 21, 2011 at 5:26 PM, Andrew Pogrebennyk
>     <andrew.pogrebennyk at portaone.com
>     <mailto:andrew.pogrebennyk at portaone.com>> wrote:
>
>         On 21.06.2011 17 <tel:21.06.2011%2017>:30, laura testi wrote:
>
>             Should I need to set a some special environment variable
>             to load ldap.so
>             in the /etc/init.d/kamailio?
>
>
>         The env output for root does not confirm my theory :) I'm not
>         sure what is preventing kamailio from communication with LDAP,
>         could it be that SELinux is enabled and denying it in some way?
>
>         -- 
>         Sincerely,
>         Andrew Pogrebennyk
>
>
>
>
>
> _______________________________________________
> sr-dev mailing list
> sr-dev at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From gminghon at hotmail.com  Wed Jun 22 12:07:28 2011
From: gminghon at hotmail.com (=?utf-8?B?4oCg4K6Qw7vil4/Rh8aD0YbOr9GI4oCgIEdvaA==?=)
Date: Wed, 22 Jun 2011 18:07:28 +0800
Subject: [SR-Users] audio issue in same nat.
In-Reply-To: <SNT101-W327D84CB88BE92529B0309CC500@phx.gbl>
References: <SNT101-W431DF375A5B660E47A0CDFCC500@phx.gbl>,
	<SNT101-W327D84CB88BE92529B0309CC500@phx.gbl>
Message-ID: <SNT101-W573B6086275BEB85D545D5CC500@phx.gbl>


oh now getting worst when i test a few more times both 101 call 102 vice versa both no audio.but 102 got audio when call voicemail.101 still no audio when call voicemail.
ReGaRds,
 
MinGh0n 

From: gminghon at hotmail.com
To: sr-users at lists.sip-router.org
Subject: audio issue in same nat.
Date: Wed, 22 Jun 2011 17:34:53 +0800












Hi List,
below is my setup..
rtpproxy and kamailio in one PC with 2 nic. (ppp0 with public IP[60.49.119.XX] and eth1 with private IP[192.168.2.3])and asterisk is on another PC with private IP[192.168.2.23]
i use realtime integration for kamailio and asterisk.http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
i have two yealink hardphone ext 101(ip 192.168.1.200) and 102(ip 192.168.1.132) and a softphone ext 103 registered successful.both hardphone are behind same nat (175.136.221.XX)and softphone ext 103(ip 10.129.138.225) behind nat also (113.210.97.XX)
ul show:database engine 'MYSQL' loadedControl engine 'FIFO' loadedentering fifo_cmd ul_dumpDomain:: location table=512 records=3 max_slot=1        AOR:: 102                Contact:: sip:102 at 175.136.221.XX:5062 Q=                        Expires:: 3109                        Callid:: 2043273564 at 175.136.221.241                        Cseq:: 4                        User-agent:: T22 7.3.0.50                        Received:: sip:175.136.221.241:1039                        State:: CS_SYNC                        Flags:: 0                        Cflag:: 192                        Socket:: udp:60.49.119.69:5060                        Methods:: 16383        AOR:: 103                Contact:: sip:103 at 113.210.97.XX:58776;transport=UDP;ob Q=                        Expires:: 294                        Callid:: oa8Pqx3mR.SVnzAVEYHTwVKZE8CbpY9l                        Cseq:: 27626                        User-agent:: v1.0.0/iPhone                        State:: CS_NEW                        Flags:: 0                        Cflag:: 0                        Socket:: udp:60.49.119.69:5060                        Methods:: 8143        AOR:: 101                Contact:: sip:101 at 175.136.221.XX:5062 Q=                        Expires:: 1738                        Callid:: 451417581 at 175.136.221.241                        Cseq:: 2                        User-agent:: T20 9.41.0.80                        State:: CS_SYNC                        Flags:: 0                        Cflag:: 0                        Socket:: udp:60.49.119.69:5060                        Methods:: 16383FIFO command was::ul_dump:openser_receiver_17783
103 try to call 102 and 101 work fine. 101 and 102 try call 103 also fine.when 101 call 102 it work fine but when 102 call 101 there is no audio for both side.102 call 101 wireshark capture on 102 sidekeep send rtp but no receive.192.168.1.132 -> 60.49.119.XX RTP
when capture on 101 side.keep send rtp but no receive.192.168.1.200 -> 60.49.119.XX RTP
and also when 101 try to call into voicemail there is no audioit keep send rtp packet but to192.168.1.200 -> 192.168.2.23 RTP
in kamailio.cfg#!WITH_NATlisten=60.49.119.XXlisten=192.168.2.3
# uncomment next line to do SIP NAT pinging                        setbflag(FLB_NATSIPPING);
nat_uac_test("19")rtpproxy -l 60.49.119.XX -s udp:127.0.0.1 is running

anyone can help me? how can i fix this?thanks in adv.

Regards, 
minghon 		 	   		  
 		 	   		   		 	   		  
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From klaus.mailinglists at pernau.at  Wed Jun 22 15:30:01 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 22 Jun 2011 15:30:01 +0200
Subject: [SR-Users] OT: are there still asymmetric SIP clients?
Message-ID: <4E01EE59.3040405@pernau.at>

Hi!

Just a small survey: Are you still using NAT detection checks? I prefer
to apply NAT traversal always (never trust the contact header) which of
course only works if all the clients are symmetric.

Thus - do you still see asymmetric SIP clients?

Thanks
Klaus


From sos at sokhapkin.dyndns.org  Wed Jun 22 15:36:00 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 22 Jun 2011 09:36:00 -0400
Subject: [SR-Users] OT: are there still asymmetric SIP clients?
In-Reply-To: <4E01EE59.3040405@pernau.at>
References: <4E01EE59.3040405@pernau.at>
Message-ID: <201106220936.00574.sos@sokhapkin.dyndns.org>

A lot of Cisco (not linksys) SIP equipment is asymmetric.

On Wednesday 22 June 2011, Klaus Darilion wrote:
> Hi!
> 
> Just a small survey: Are you still using NAT detection checks? I prefer
> to apply NAT traversal always (never trust the contact header) which of
> course only works if all the clients are symmetric.
> 
> Thus - do you still see asymmetric SIP clients?
> 
> Thanks
> 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 klaus.mailinglists at pernau.at  Wed Jun 22 15:54:23 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 22 Jun 2011 15:54:23 +0200
Subject: [SR-Users] OT: are there still asymmetric SIP clients?
In-Reply-To: <201106220936.00574.sos@sokhapkin.dyndns.org>
References: <4E01EE59.3040405@pernau.at>
	<201106220936.00574.sos@sokhapkin.dyndns.org>
Message-ID: <4E01F40F.2060405@pernau.at>

So what do you use as nat_check? Only the received IP address?

regards
Klaus

Am 22.06.2011 15:36, schrieb Sergey Okhapkin:
> A lot of Cisco (not linksys) SIP equipment is asymmetric.
> 
> On Wednesday 22 June 2011, Klaus Darilion wrote:
>> Hi!
>>
>> Just a small survey: Are you still using NAT detection checks? I prefer
>> to apply NAT traversal always (never trust the contact header) which of
>> course only works if all the clients are symmetric.
>>
>> Thus - do you still see asymmetric SIP clients?
>>
>> Thanks
>> 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 sos at sokhapkin.dyndns.org  Wed Jun 22 16:00:12 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 22 Jun 2011 10:00:12 -0400
Subject: [SR-Users] OT: are there still asymmetric SIP clients?
In-Reply-To: <4E01F40F.2060405@pernau.at>
References: <4E01EE59.3040405@pernau.at>
	<201106220936.00574.sos@sokhapkin.dyndns.org>
	<4E01F40F.2060405@pernau.at>
Message-ID: <201106221000.12306.sos@sokhapkin.dyndns.org>

I use a separate IP address to accept calls from such SIP clients and do not 
check for NAT on those calls.

On Wednesday 22 June 2011, Klaus Darilion wrote:
> So what do you use as nat_check? Only the received IP address?
> 
> regards
> Klaus
> 
> Am 22.06.2011 15:36, schrieb Sergey Okhapkin:
> > A lot of Cisco (not linksys) SIP equipment is asymmetric.
> > 
> > On Wednesday 22 June 2011, Klaus Darilion wrote:
> >> Hi!
> >> 
> >> Just a small survey: Are you still using NAT detection checks? I prefer
> >> to apply NAT traversal always (never trust the contact header) which of
> >> course only works if all the clients are symmetric.
> >> 
> >> Thus - do you still see asymmetric SIP clients?
> >> 
> >> Thanks
> >> 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
> 
> _______________________________________________
> 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 Jun 22 18:51:42 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 22 Jun 2011 18:51:42 +0200
Subject: [SR-Users] Ukulele Kamailio Jingle
Message-ID: <4E021D9E.7000407@gmail.com>

Hello,

a good friend, Javi Beltran, recorded a composition playing Ukulele 
(aka, ukelele - Hawaiian instrument - 
http://en.wikipedia.org/wiki/Ukulele). The jingle is available at:
   * http://asipto.com/u/39

If you like it, feel free to use it in relation with Kamailio project. 
Javi's compositions are open sourced, available at:
   * http://www.javibeltran.com/?lang=en

I posted a news about it on project's site:
   * http://www.kamailio.org/w/2011/06/ukulele-kamailio-jingle/

If you or your friends have done any artistic work (music, graphics, 
a.s.o.) related to Kamailio, feel free to send it over and I will make 
them available on project's site.

Thanks,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From oej at edvina.net  Wed Jun 22 21:57:57 2011
From: oej at edvina.net (Olle E Johansson)
Date: Wed, 22 Jun 2011 21:57:57 +0200
Subject: [SR-Users] Kamailio/Asterisk/FreeSWitch dinner in Madrid - Thursday
	next week
Message-ID: <D39AE593-B353-4A2E-A169-6A99363A8367@edvina.net>

Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - The SIP master class.

Maybe we can organize a voip nerd dinner on Thursday evening? If you're interested, please e-mail me off list and I'll send out more details later. 

Greetings
/Olle

PS. E-mail off list :-)

From ibc at aliax.net  Thu Jun 23 01:45:46 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Thu, 23 Jun 2011 01:45:46 +0200
Subject: [SR-Users] OT: are there still asymmetric SIP clients?
In-Reply-To: <4E01EE59.3040405@pernau.at>
References: <4E01EE59.3040405@pernau.at>
Message-ID: <BANLkTi=8=p=GT0engm8Yz534QjpS2i6Kow@mail.gmail.com>

2011/6/22 Klaus Darilion <klaus.mailinglists at pernau.at>:
> Just a small survey: Are you still using NAT detection checks? I prefer
> to apply NAT traversal always (never trust the contact header) which of
> course only works if all the clients are symmetric.

I would like to tell a recent issue I've suffered in my company:

- Kamailio with public IP and optional rtpproxy server.
- Two users behind same NAT.
- Both users use STUN so announce public IP:ports in headers and SDP.
- User A calls to B.
- Kamailio does not detect NAT (due to STUN) so doesn't apply
rtpproxy, neither when routes the call to user B.
- So user A sends RTP to the public IP of its NAT router, same as user B.
- In some routers this work, but in this case it does not properly
perform NAT port forwarding and there is no audio.

Solutions:
1) Remove STUN from clients (not always feasible).
2) Always use RtpProxy. Easier.



> Thus - do you still see asymmetric SIP clients?

Yes, I've seen some Alcatel OXO using asymmetric SIP. A pain if you do
"force_rport".

Solution:
- Just perform "force_rport" if the client is detected as behind NAT
(note that assymetric SIP clients can ***NEVER*** run behind NAT).


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


From omar at 321communications.com  Thu Jun 23 03:01:58 2011
From: omar at 321communications.com (Omar)
Date: Wed, 22 Jun 2011 21:01:58 -0400
Subject: [SR-Users] Carrierroute Compilation error Kamailio 3.1.4
In-Reply-To: <4E019E38.1050407@1and1.ro>
References: <0FFF2239-59E1-4032-B18E-84A35289EDBA@321communications.com>	<201106201111.11708.henning.westerholt@1und1.de>	<8869B5C4-D4D4-4B06-9015-7E753BC68E6C@321communications.com>	<201106211730.14087.henning.westerholt@1und1.de>
	<9FB43C00-92BF-445D-970F-CC7245D4E501@321communications.com>
	<4E019E38.1050407@1and1.ro>
Message-ID: <CEA50C62-83E4-4E13-BE6F-58415D7376AC@321communications.com>

great that fixed the carrierroute even for some reason i got a several warning:

make[1]: confuse-config: Command not found


But now i am stuck in the MYSQL


On Jun 22, 2011, at 3:48 AM, marius zbihlei wrote:

> On 06/22/2011 03:41 AM, Omar wrote:
>> Here we go
>> 
>>   
> 
> Hello Omar,
> 
> You are missing a -DOPENSER_MOD_INTERFACE in the gcc command line. This is set in modules/carrierroute/Makefile  like below
> 
> ...
> DEFS+=$(CONFUSEDEFS)
> LIBS=$(CONFUSELIBS)
> 
> DEFS+=-DOPENSER_MOD_INTERFACE
> 
> SERLIBPATH=../../lib
> SER_LIBS+=$(SERLIBPATH)/trie/trie $(SERLIBPATH)/srdb1/srdb1 $(SERLIBPATH)/kmi/kmi $(SERLIBPATH)/kcore/kcore
> 
> include ../../Makefile.modules
> ....
> 
> (This is just under the libconfuse find).
> Can you check that this Makefile is unchanged(or at least you have this line). Also try to run make QUIET=0 from the modules/carrierroute directory and see what -D flags are passed to the compiler
> 
> Thanks
> Marius
>> bash-3.00# make clean
>> bash-3.00# make quiet=0
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c prime_hash.c -o prime_hash.o -MMD -MP
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c cr_data.c -o cr_data.o -MMD -MP
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -I/opt/openser/include -DMOD_NAME='"carrierroute"' -c carrierroute.c -o carrierroute.o -MMD -MP
>> carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a function)
>> carrierroute.c:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cmds'
>> carrierroute.c:132: error: variable 'exports' has initializer but incomplete type
>> carrierroute.c:133: warning: excess elements in struct initializer
>> carrierroute.c:133: warning: (near initialization for 'exports')
>> carrierroute.c:134: warning: excess elements in struct initializer
>> carrierroute.c:134: warning: (near initialization for 'exports')
>> carrierroute.c:135: error: 'cmds' undeclared here (not in a function)
>> carrierroute.c:135: warning: excess elements in struct initializer
>> carrierroute.c:135: warning: (near initialization for 'exports')
>> carrierroute.c:136: warning: excess elements in struct initializer
>> carrierroute.c:136: warning: (near initialization for 'exports')
>> carrierroute.c:137: warning: excess elements in struct initializer
>> carrierroute.c:137: warning: (near initialization for 'exports')
>> carrierroute.c:138: warning: excess elements in struct initializer
>> carrierroute.c:138: warning: (near initialization for 'exports')
>> carrierroute.c:139: warning: excess elements in struct initializer
>> carrierroute.c:139: warning: (near initialization for 'exports')
>> carrierroute.c:140: warning: excess elements in struct initializer
>> carrierroute.c:140: warning: (near initialization for 'exports')
>> carrierroute.c:141: warning: excess elements in struct initializer
>> carrierroute.c:141: warning: (near initialization for 'exports')
>> carrierroute.c:142: warning: excess elements in struct initializer
>> carrierroute.c:142: warning: (near initialization for 'exports')
>> carrierroute.c:143: warning: excess elements in struct initializer
>> carrierroute.c:143: warning: (near initialization for 'exports')
>> carrierroute.c:145: warning: excess elements in struct initializer
>> carrierroute.c:145: warning: (near initialization for 'exports')
>> carrierroute.c: In function 'mod_init':
>> carrierroute.c:161: error: invalid use of undefined type 'struct module_exports'
>> make: *** [carrierroute.o] Error 1
>> bash-3.00#
>> 
>> On Jun 21, 2011, at 11:30 AM, Henning Westerholt wrote:
>> 
>>   
>>> On Monday 20 June 2011, Omar wrote:
>>>     
>>>> Everything compiles fine until the showed error the were errors about the
>>>> libconfuse which I fixed stating the path to the library, yes lib confuse
>>>> is installed I am using the same one I installed for the kamailio 1.4.3.
>>>>       
>>> Hi Omar,
>>> 
>>> ok, strange. If it works for the 1.4.3 it should also work for the new
>>> version, at least i'm not aware of any changes in this regards. Can you please
>>> provide the output of a "make clean; make quiet=0" command in the
>>> modules/carrierroute directory?
>>> 
>>> Best regards,
>>> 
>>> Henning
>>> 
>>>     
>> 
>> _______________________________________________
>> 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 omar at 321communications.com  Thu Jun 23 03:03:44 2011
From: omar at 321communications.com (Omar)
Date: Wed, 22 Jun 2011 21:03:44 -0400
Subject: [SR-Users] MYSQL Compilation Kamailio 3.1.4
Message-ID: <21FAF94F-ADCB-4714-8154-3D43E1308D0D@321communications.com>

bash-3.00# make clean
bash-3.00# make quiet=0 
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c km_db_mysql.c -o km_db_mysql.o -MMD -MP
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_uri.c -o my_uri.o -MMD -MP
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_res.c -o my_res.o -MMD -MP
/opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_cmd.c -o my_cmd.o -MMD -MP
In file included from ../../globals.h:37,
                 from ../../pt.h:42,
                 from ../../counters.h:43,
                 from mysql_mod.h:39,
                 from my_cmd.c:40:
../../ip_addr.h:86: error: field 'sin6' has incomplete type
../../ip_addr.h: In function 'ip_addr_loopback':
../../ip_addr.h:283: warning: implicit declaration of function 'htonl'
../../ip_addr.h:286: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK'
../../ip_addr.h: In function 'sockaddr2ip_addr':
../../ip_addr.h:353: error: dereferencing pointer to incomplete type
../../ip_addr.h: In function 'su_getport':
../../ip_addr.h:399: warning: implicit declaration of function 'ntohs'
../../ip_addr.h: In function 'su_setport':
../../ip_addr.h:418: warning: implicit declaration of function 'htons'
make: *** [my_cmd.o] Error 1


the onyl thing i did is uncomment the CROSSCOMPILE line 

regards

O

From omar at 321communications.com  Thu Jun 23 03:05:36 2011
From: omar at 321communications.com (Omar)
Date: Wed, 22 Jun 2011 21:05:36 -0400
Subject: [SR-Users] Don't publish Is not supposed the e-mail should be Hiden
Message-ID: <E30A9B5D-27C5-45A6-A265-503064EA9D93@321communications.com>

I really don't want to share my e-mail publicly, i remember there was an option before.

O

From roman at dmytriv.com  Thu Jun 23 03:28:37 2011
From: roman at dmytriv.com (Roman Dmytriv)
Date: Wed, 22 Jun 2011 21:28:37 -0400
Subject: [SR-Users] apt-get install doesn't work
Message-ID: <001901cc3144$e0dfc400$a29f4c00$@com>

Hello,

 

I was trying to install kamailio using your instruction from
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour

 

I run Debian Squeeze. Imported apt repository keys, created this file
</etc/apt/sources.list.d/kamailio.list> with this information in it <deb
http://deb.kamailio.org/kamailio31-nightly squeeze main>
 
Now when I run apt-get install kamailio kamailio-mysql-modules
kamailio-tls-modules I got this response: 

 

root at s2:/etc/apt/sources.list.d# apt-get install kamailio
kamailio-mysql-modules kamailio-tls-modules

Reading package lists... Done

Building dependency tree       

Reading state information... Done

E: Unable to locate package kamailio

E: Unable to locate package kamailio-mysql-modules

E: Unable to locate package kamailio-tls-modules

 

Can you give me a hint what could be wrong?

 

 

Regards

 

 

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From abalashov at evaristesys.com  Thu Jun 23 03:30:41 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 22 Jun 2011 21:30:41 -0400
Subject: [SR-Users] apt-get install doesn't work
In-Reply-To: <001901cc3144$e0dfc400$a29f4c00$@com>
References: <001901cc3144$e0dfc400$a29f4c00$@com>
Message-ID: <4E029741.9020709@evaristesys.com>

Roman,

On 06/22/2011 09:28 PM, Roman Dmytriv wrote:

> root at s2:/etc/apt/sources.list.d#
> <mailto:root at s2:/etc/apt/sources.list.d#> apt-get install kamailio
> kamailio-mysql-modules kamailio-tls-modules
>
> Reading package lists... Done
>
> Building dependency tree
>
> Reading state information... Done
>
> E: Unable to locate package kamailio
>
> E: Unable to locate package kamailio-mysql-modules
>
> E: Unable to locate package kamailio-tls-modules
>
> Can you give me a hint what could be wrong?

Can you try this and paste the output?

apt-cache search kamailio | grep -i '^kamailio'

?

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From roman at dmytriv.com  Thu Jun 23 03:34:49 2011
From: roman at dmytriv.com (Roman Dmytriv)
Date: Wed, 22 Jun 2011 21:34:49 -0400
Subject: [SR-Users] apt-get install doesn't work
In-Reply-To: <4E029741.9020709@evaristesys.com>
References: <001901cc3144$e0dfc400$a29f4c00$@com>
	<4E029741.9020709@evaristesys.com>
Message-ID: <002701cc3145$bf43d560$3dcb8020$@com>


-----Original Message-----
From: Alex Balashov [mailto:abalashov at evaristesys.com] 
Sent: Wednesday, June 22, 2011 9:31 PM
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] apt-get install doesn't work

Roman,

On 06/22/2011 09:28 PM, Roman Dmytriv wrote:

> root at s2:/etc/apt/sources.list.d#
> <mailto:root at s2:/etc/apt/sources.list.d#> apt-get install kamailio
> kamailio-mysql-modules kamailio-tls-modules
>
> Reading package lists... Done
>
> Building dependency tree
>
> Reading state information... Done
>
> E: Unable to locate package kamailio
>
> E: Unable to locate package kamailio-mysql-modules
>
> E: Unable to locate package kamailio-tls-modules
>
> Can you give me a hint what could be wrong?

Can you try this and paste the output?

apt-cache search kamailio | grep -i '^kamailio'

?

Nothing came out

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.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 abalashov at evaristesys.com  Thu Jun 23 03:35:55 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 22 Jun 2011 21:35:55 -0400
Subject: [SR-Users] apt-get install doesn't work
In-Reply-To: <002701cc3145$bf43d560$3dcb8020$@com>
References: <001901cc3144$e0dfc400$a29f4c00$@com>	<4E029741.9020709@evaristesys.com>
	<002701cc3145$bf43d560$3dcb8020$@com>
Message-ID: <4E02987B.4050400@evaristesys.com>

On 06/22/2011 09:34 PM, Roman Dmytriv wrote:
>
> -----Original Message-----
> From: Alex Balashov [mailto:abalashov at evaristesys.com]
> Sent: Wednesday, June 22, 2011 9:31 PM
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] apt-get install doesn't work
>
> Roman,
>
> On 06/22/2011 09:28 PM, Roman Dmytriv wrote:
>
>> root at s2:/etc/apt/sources.list.d#
>> <mailto:root at s2:/etc/apt/sources.list.d#>  apt-get install kamailio
>> kamailio-mysql-modules kamailio-tls-modules
>>
>> Reading package lists... Done
>>
>> Building dependency tree
>>
>> Reading state information... Done
>>
>> E: Unable to locate package kamailio
>>
>> E: Unable to locate package kamailio-mysql-modules
>>
>> E: Unable to locate package kamailio-tls-modules
>>
>> Can you give me a hint what could be wrong?
>
> Can you try this and paste the output?
>
> apt-cache search kamailio | grep -i '^kamailio'
>
> ?
>
> Nothing came out
>

Then something did not work with adding the Kamailio repository.  Did 
you run 'apt-get update' before attempting to 'apt-get install' the 
packages?

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From roman at dmytriv.com  Thu Jun 23 03:38:49 2011
From: roman at dmytriv.com (Roman Dmytriv)
Date: Wed, 22 Jun 2011 21:38:49 -0400
Subject: [SR-Users] apt-get install doesn't work
In-Reply-To: <4E02987B.4050400@evaristesys.com>
References: <001901cc3144$e0dfc400$a29f4c00$@com>	<4E029741.9020709@evaristesys.com>	<002701cc3145$bf43d560$3dcb8020$@com>
	<4E02987B.4050400@evaristesys.com>
Message-ID: <002a01cc3146$4de7a440$e9b6ecc0$@com>



-----Original Message-----
From: Alex Balashov [mailto:abalashov at evaristesys.com] 
Sent: Wednesday, June 22, 2011 9:36 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Subject: Re: [SR-Users] apt-get install doesn't work

On 06/22/2011 09:34 PM, Roman Dmytriv wrote:
>
> -----Original Message-----
> From: Alex Balashov [mailto:abalashov at evaristesys.com]
> Sent: Wednesday, June 22, 2011 9:31 PM
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] apt-get install doesn't work
>
> Roman,
>
> On 06/22/2011 09:28 PM, Roman Dmytriv wrote:
>
>> root at s2:/etc/apt/sources.list.d#
>> <mailto:root at s2:/etc/apt/sources.list.d#>  apt-get install kamailio
>> kamailio-mysql-modules kamailio-tls-modules
>>
>> Reading package lists... Done
>>
>> Building dependency tree
>>
>> Reading state information... Done
>>
>> E: Unable to locate package kamailio
>>
>> E: Unable to locate package kamailio-mysql-modules
>>
>> E: Unable to locate package kamailio-tls-modules
>>
>> Can you give me a hint what could be wrong?
>
> Can you try this and paste the output?
>
> apt-cache search kamailio | grep -i '^kamailio'
>
> ?
>
> Nothing came out
>

Then something did not work with adding the Kamailio repository.  Did 
you run 'apt-get update' before attempting to 'apt-get install' the 
packages?

That was it! Thank you Alex!!!

Roman

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.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 abalashov at evaristesys.com  Thu Jun 23 03:40:25 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Wed, 22 Jun 2011 21:40:25 -0400
Subject: [SR-Users] apt-get install doesn't work
In-Reply-To: <002a01cc3146$4de7a440$e9b6ecc0$@com>
References: <001901cc3144$e0dfc400$a29f4c00$@com>	<4E029741.9020709@evaristesys.com>	<002701cc3145$bf43d560$3dcb8020$@com>	<4E02987B.4050400@evaristesys.com>
	<002a01cc3146$4de7a440$e9b6ecc0$@com>
Message-ID: <4E029989.3010805@evaristesys.com>

On 06/22/2011 09:38 PM, Roman Dmytriv wrote:

> That was it! Thank you Alex!!!

Naturally.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From brett at woollum.com  Thu Jun 23 07:10:07 2011
From: brett at woollum.com (Brett Woollum)
Date: Wed, 22 Jun 2011 22:10:07 -0700 (PDT)
Subject: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP
	endpoint
In-Reply-To: <1921601578.20711308805388318.JavaMail.root@mail>
Message-ID: <277720305.20731308805807198.JavaMail.root@mail>


Hello, 

We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. Everything seems to be working well, until today. 

I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. 

If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. 

What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. 

Thanks! 
Brett 
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From abalashov at evaristesys.com  Thu Jun 23 07:22:18 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 23 Jun 2011 01:22:18 -0400
Subject: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to
	SIP endpoint
In-Reply-To: <277720305.20731308805807198.JavaMail.root@mail>
References: <277720305.20731308805807198.JavaMail.root@mail>
Message-ID: <1A4D664E-CA7F-4E29-A109-D56D0FCCD9CC@evaristesys.com>

This is a complex topic.  There is no way for a proxy like Kamailio to detect this scenario per se.  Kamailio reacts to and forwards signaling events.  If an endpoint disappears, it won't send any of those to indicate that it has gone away.  How would Kamailio know?  Media stream timeout?  Kamailio doesn't relay media.

Your Kamailio-side solution is a dialog timeout, requiring use of dialog-stateful tracking using the dialog module.  But that will time out calls indiscriminately, so you need to make it long enough to not anger your users but short enough to be useful.

Your endpoint solution is SIP Session Timers.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett at woollum.com> wrote:

> Hello,
> 
> We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. Everything seems to be working well, until today.
> 
> I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related.
> 
> If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt.
> 
> What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have.
> 
> Thanks!
> Brett
> _______________________________________________
> 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 cargius at gmail.com  Thu Jun 23 08:35:59 2011
From: cargius at gmail.com (Giuseppe Carella)
Date: Thu, 23 Jun 2011 08:35:59 +0200
Subject: [SR-Users]  Dispatcher module for INVITE
Message-ID: <BANLkTi=ToWS5FHmDx=8RK5Xpxic8-x--2w@mail.gmail.com>

Hi all,

I'm using kamailio dispatcher module between
the OpenIMSCore and sailfin. Normally, without the dispatcher,
all works fine. From the IMS Bench I can see that there are
many successful call, but when I use the dispatcher module
I'm having a lot of retransmission in the first part of the INVITE
protocol. I'm using the default configuration file:

# ----------------- setting module-specific parameters ---------------
# -- dispatcher params --

modparam("dispatcher", "list_file", "../etc/dispatcher.list")
# modparam("dispatcher", "force_dst", 1)

route{
if ( !mf_process_maxfwd_header("10") )
{
 sl_send_reply("483","To Many Hops");
 drop();
};
 ds_select_dst("1", "0");
 forward();
# t_relay();
}


I tried also the t_relay() function instead of the forward
command, but the problem is the same.

Thanks in advance.

Bests,

Giuseppe
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From brett at woollum.com  Thu Jun 23 08:39:08 2011
From: brett at woollum.com (Brett Woollum)
Date: Wed, 22 Jun 2011 23:39:08 -0700 (PDT)
Subject: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to
 SIP endpoint
In-Reply-To: <452054889.20811308811140469.JavaMail.root@mail>
Message-ID: <901861315.20831308811148354.JavaMail.root@mail>

Hi Alex, 

Thanks for this information. I've started researching the session-timer capabilities in Asterisk, and I think that's my solution. I've already implemented it on a test system and it works well, except that it's using reINVITES to update as opposed to UPDATE messages, resulting in chops in the audio every so often. I'll research this further though. 

Thanks again! 
Brett 

----- Original Message ----- 
From: "Alex Balashov" <abalashov at evaristesys.com> 
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users at lists.sip-router.org> 
Cc: sr-users at lists.sip-router.org 
Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific 
Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint 


This is a complex topic. There is no way for a proxy like Kamailio to detect this scenario per se. Kamailio reacts to and forwards signaling events. If an endpoint disappears, it won't send any of those to indicate that it has gone away. How would Kamailio know? Media stream timeout? Kamailio doesn't relay media. 


Your Kamailio-side solution is a dialog timeout, requiring use of dialog-stateful tracking using the dialog module. But that will time out calls indiscriminately, so you need to make it long enough to not anger your users but short enough to be useful. 


Your endpoint solution is SIP Session Timers. 

-- 
Alex Balashov - Principal 
Evariste Systems LLC 
260 Peachtree Street NW 
Suite 2200 
Atlanta, GA 30303 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/ 

On Jun 23, 2011, at 1:10 AM, Brett Woollum < brett at woollum.com > wrote: 







Hello, 

We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb . Everything seems to be working well, until today. 

I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. 

If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. 

What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. 

Thanks! 
Brett 



_______________________________________________ 
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 
-------------- next part --------------
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From carsten at ng-voice.com  Thu Jun 23 09:46:11 2011
From: carsten at ng-voice.com (Carsten Bock)
Date: Thu, 23 Jun 2011 09:46:11 +0200
Subject: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to
 SIP endpoint
In-Reply-To: <901861315.20831308811148354.JavaMail.root@mail>
References: <452054889.20811308811140469.JavaMail.root@mail>
	<901861315.20831308811148354.JavaMail.root@mail>
Message-ID: <BANLkTinxb3O2T3zO_7jEF+mivoWGPuHBHw@mail.gmail.com>

Hi,

another solution might be, to either configure an RTP-Timeout on the
Asterisk (if you send your calls through the asterisk anyway).
You might also consider using the RTPProxy with the patch in the
sip-router-repository. With the patch, the RTPProxy will trigger a
teardown of calls (via XML-RPC) if the RTP-Session has a timeout.

Carsten

2011/6/23 Brett Woollum <brett at woollum.com>:
> Hi Alex,
>
> Thanks for this information. I've started researching the session-timer
> capabilities in Asterisk, and I think that's my solution. I've already
> implemented it on a test system and it works well, except that it's using
> reINVITES to update as opposed to UPDATE messages, resulting in chops in the
> audio every so often. I'll research this further though.
>
> Thanks again!
> Brett
>
> ----- Original Message -----
> From: "Alex Balashov" <abalashov at evaristesys.com>
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users
> Mailing List" <sr-users at lists.sip-router.org>
> Cc: sr-users at lists.sip-router.org
> Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific
> Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to
> SIP endpoint
>
> This is a complex topic. ?There is no way for a proxy like Kamailio to
> detect this scenario per se. ?Kamailio reacts to and forwards signaling
> events. ?If an endpoint disappears, it won't send any of those to indicate
> that it has gone away. ?How would Kamailio know? ?Media stream timeout?
> ?Kamailio doesn't relay media.
> Your Kamailio-side solution is a dialog timeout, requiring use of
> dialog-stateful tracking using the dialog module. ?But that will time out
> calls indiscriminately, so you need to make it long enough to not anger your
> users but short enough to be useful.
> Your endpoint solution is SIP Session Timers.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
> On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett at woollum.com> wrote:
>
> Hello,
>
> We are running Kamailio as a registration point for our SIP phones, which
> then interacts with Asterisk. SIP registrations are processed by Kamailio,
> but everything else is passed to Asterisk. The Kamailio configuration is
> close to the article at:
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
> Everything seems to be working well, until today.
>
> I found several calls today that were still connected to our provider, even
> though our SIP phones were not active. There were three calls with timers at
> 9 hours and counting. We had some IP connectivity issues earlier today, and
> I'm wonder if it's related.
>
> If a SIP phone was connected and on a call (through kamailio), and the
> kamailio/asterisk servers became unreachable, the SIP phones will drop the
> call. But, it appears that kamailio/asterisk never drop the call in this
> case, and the call stays live with the carrier. I had to manually kill the
> calls by command prompt.
>
> What's the best way to handle this? Is there a way to have kamailio or
> asterisk poll the phone to see if it's still on the call or something? How
> can I give visibility to asterisk or kamailio so the calls are always
> dropped properly? I don't want to run up a large bill because of calls that
> didn't terminate when they should have.
>
> Thanks!
> Brett
>
> _______________________________________________
> 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
> _______________________________________________
> 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
http://www.ng-voice.com
mailto:carsten at ng-voice.com

Schomburgstr. 80
22767 Hamburg
Germany

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220


From brett at woollum.com  Thu Jun 23 09:49:00 2011
From: brett at woollum.com (Brett Woollum)
Date: Thu, 23 Jun 2011 00:49:00 -0700 (PDT)
Subject: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to
 SIP endpoint
In-Reply-To: <291259782.21041308815304468.JavaMail.root@mail>
Message-ID: <937899488.21061308815340501.JavaMail.root@mail>

Hi Carsten, 

Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. 

I will look into that patch. 

Thanks! 


Brett 

----- Original Message ----- 
From: "Carsten Bock" <carsten at ng-voice.com> 
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users at lists.sip-router.org> 
Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint 

Hi, 

another solution might be, to either configure an RTP-Timeout on the 
Asterisk (if you send your calls through the asterisk anyway). 
You might also consider using the RTPProxy with the patch in the 
sip-router-repository. With the patch, the RTPProxy will trigger a 
teardown of calls (via XML-RPC) if the RTP-Session has a timeout. 

Carsten 

2011/6/23 Brett Woollum <brett at woollum.com>: 
> Hi Alex, 
> 
> Thanks for this information. I've started researching the session-timer 
> capabilities in Asterisk, and I think that's my solution. I've already 
> implemented it on a test system and it works well, except that it's using 
> reINVITES to update as opposed to UPDATE messages, resulting in chops in the 
> audio every so often. I'll research this further though. 
> 
> Thanks again! 
> Brett 
> 
> ----- Original Message ----- 
> From: "Alex Balashov" <abalashov at evaristesys.com> 
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users 
> Mailing List" <sr-users at lists.sip-router.org> 
> Cc: sr-users at lists.sip-router.org 
> Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific 
> Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to 
> SIP endpoint 
> 
> This is a complex topic. There is no way for a proxy like Kamailio to 
> detect this scenario per se. Kamailio reacts to and forwards signaling 
> events. If an endpoint disappears, it won't send any of those to indicate 
> that it has gone away. How would Kamailio know? Media stream timeout? 
> Kamailio doesn't relay media. 
> Your Kamailio-side solution is a dialog timeout, requiring use of 
> dialog-stateful tracking using the dialog module. But that will time out 
> calls indiscriminately, so you need to make it long enough to not anger your 
> users but short enough to be useful. 
> Your endpoint solution is SIP Session Timers. 
> 
> -- 
> Alex Balashov - Principal 
> Evariste Systems LLC 
> 260 Peachtree Street NW 
> Suite 2200 
> Atlanta, GA 30303 
> Tel: +1-678-954-0670 
> Fax: +1-404-961-1892 
> Web: http://www.evaristesys.com/ 
> On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett at woollum.com> wrote: 
> 
> Hello, 
> 
> We are running Kamailio as a registration point for our SIP phones, which 
> then interacts with Asterisk. SIP registrations are processed by Kamailio, 
> but everything else is passed to Asterisk. The Kamailio configuration is 
> close to the article at: 
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. 
> Everything seems to be working well, until today. 
> 
> I found several calls today that were still connected to our provider, even 
> though our SIP phones were not active. There were three calls with timers at 
> 9 hours and counting. We had some IP connectivity issues earlier today, and 
> I'm wonder if it's related. 
> 
> If a SIP phone was connected and on a call (through kamailio), and the 
> kamailio/asterisk servers became unreachable, the SIP phones will drop the 
> call. But, it appears that kamailio/asterisk never drop the call in this 
> case, and the call stays live with the carrier. I had to manually kill the 
> calls by command prompt. 
> 
> What's the best way to handle this? Is there a way to have kamailio or 
> asterisk poll the phone to see if it's still on the call or something? How 
> can I give visibility to asterisk or kamailio so the calls are always 
> dropped properly? I don't want to run up a large bill because of calls that 
> didn't terminate when they should have. 
> 
> Thanks! 
> Brett 
> 
> _______________________________________________ 
> 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 
> _______________________________________________ 
> 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 
http://www.ng-voice.com 
mailto:carsten at ng-voice.com 

Schomburgstr. 80 
22767 Hamburg 
Germany 

Mobile +49 179 2021244 
Office +49 40 34927219 
Fax +49 40 34927220 

_______________________________________________ 
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 dotnetdub at gmail.com  Thu Jun 23 18:58:50 2011
From: dotnetdub at gmail.com (dotnetdub)
Date: Thu, 23 Jun 2011 17:58:50 +0100
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <4DFB8D4F.5070506@evaristesys.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
	<4DFB8D4F.5070506@evaristesys.com>
Message-ID: <BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>

On 17 June 2011 18:22, Alex Balashov <abalashov at evaristesys.com> wrote:

> Brian,
>
> Remote-Party-ID is a long-obsolete draft that never became an RFC; why are
> you still using it?
>
> Anyway, if you know the 'privacy' header parameter will always be there,
> you can do this, assuming is_present_hf("Remote-Party-**ID"), of course.
>
>   $var(x) = $(hdr(Remote-Party-ID){s.**select,1,>});
>
>   xlog("L_INFO", "Remote Party ID privacy value:
> $(var(x){param.value,privacy})**\n");
>
>

Hi Alex,

This doesn't seem to be working for me..

/sbin/kamailio[25291]: INFO: <script>: Remote Party ID privacy value:



From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.2.1308848361.2669.sr-users@lists.sip-router.org>

;privacy=full;screen=no>

Let me know if I'm doing something wrong.

Thanks
Brian.






> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.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>
>

--000e0cd2e0a607f1d104a66401de
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On 17 June 2011 18:22, Alex Balashov <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:abalashov at evaristesys.com">abalashov at e=
varistesys.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

Brian,<br>
<br>
Remote-Party-ID is a long-obsolete draft that never became an RFC; why are =
you still using it?<br>
<br>
Anyway, if you know the &#39;privacy&#39; header parameter will always be t=
here, you can do this, assuming is_present_hf(&quot;Remote-Party-<u></u>ID&=
quot;), of course.<br>
<br>
 =A0 $var(x) =3D $(hdr(Remote-Party-ID){s.<u></u>select,1,&gt;});<br>
<br>
 =A0 xlog(&quot;L_INFO&quot;, &quot;Remote Party ID privacy value: $(var(x)=
{param.value,privacy})<u></u>\n&quot;);<br>
<br></blockquote><div><br></div><div><br></div><div>Hi Alex,</div><div><br>=
</div><div>This doesn&#39;t seem to be working for me..</div><div><br></div=
><div><div>/sbin/kamailio[25291]: INFO: &lt;script&gt;: Remote Party ID pri=
vacy value:=A0</div>

</div><div><br></div><div><br></div><div>From the initial invite:=A0Remote-=
Party-ID: &lt;<a href=3D"mailto:sip%3A1234567 at domain.ie">sip:1234567 at domain=
.ie</a>;privacy=3Dfull;screen=3Dno&gt;</div><div><br></div><div>Let me know=
 if I&#39;m doing something wrong.</div>

<div><br></div><div>Thanks</div><div>Brian.</div><div><br></div><div><br></=
div><div><br></div><div><br></div><div>=A0</div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex;">


-- <br>
Alex Balashov - Principal<br>
Evariste Systems LLC<br>
260 Peachtree Street NW<br>
Suite 2200<br>
Atlanta, GA 30303<br>
Tel: +1-678-954-0670<br>
Fax: +1-404-961-1892<br>
Web: <a href=3D"http://www.evaristesys.com/" target=3D"_blank">http://www.e=
varistesys.com/</a><br>
<br>
______________________________<u></u>_________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href=3D"mailto:sr-users at lists.sip-router.org" target=3D"_blank">sr-users=
@lists.sip-router.org</a><br>
<a href=3D"http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" t=
arget=3D"_blank">http://lists.sip-router.org/<u></u>cgi-bin/mailman/listinf=
o/sr-<u></u>users</a><br>
</blockquote></div><br>

--000e0cd2e0a607f1d104a66401de--


From miconda at gmail.com  Thu Jun 23 19:20:31 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Thu, 23 Jun 2011 19:20:31 +0200
Subject: [SR-Users] an idea - the SIP browser - elsipo
Message-ID: <4E0375DF.4090703@gmail.com>

Hello,

I want to present an idea and see if it is something people find 
benefits with such approach.

Some time ago, due to existence of only SIP client applications (aka 
softphones) that have static GUI, which were designed for generic 
purposes, do not present what servces the currently used provider offers 
nor are easy to customize, I played with the idea of a SIP application 
with the GUI in HTML/CSS/Javascript.

Being busy with Kamailio, the whole thing was postponed, but during last 
days I took the time and updated the code to latest developing tools 
(QT/Webkit and pjsip) and made it public on github, see:
   * http://asipto.com/u/3c

A screenshot and some details in the wiki at:
   * http://asipto.com/u/3b

The UI can be pulled from a remote site via http/https for example. The 
approach was to have a framework to render the UI, used QT/Webkit for 
that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to 
Javascript. I worked with pjsip, but the system is plugin based, so 
pjsip plugin can be replaced, plugins for other things (not only 
sip/voip) can be written easily.

The code is just for demo/prototyping, with the goal of showing 
feasibility. At this moment can be used to make/receive calls and 
send/receive instant messaging. In addition, content of UI can be 
carried by SIP messages - the examples added so far is to request some 
server statistics when clicking on a link of UI and the SIP server 
pushing periodically some info to a section of UI.

Personally I don't have any plan to go into client side development, 
still the idea was too appealing to try and I thought would give lot of 
value and enable new features on SIP services to make user experience 
more pleasant.

Therefore the target is to see if any of you think is something that 
worth pursuing and if yes, get people on board to develop it further 
from prototype phase.

In the wiki I listed a part of my wish list with current VoIP/SIP 
services that would be trivial to implement with this approach. The 
naming as SIP browser instead of softphone hopefully suggest also the 
purpose of the application - to be able to browse the services offered 
by provider at that time and the provider to be able to push content 
back to application (on demand or periodically). Development was done on 
Mac OS X, a tarball with binaries for this OS is available for download 
on github - compiling for other OSes should go pretty straightforward.

At the moment the idea poped up there was no such approach that I could 
find (at least not in open source space), did I miss something, are you 
aware of any similar approach? What is your opinion overall?

Thanks,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From skchopperguy at gmail.com  Thu Jun 23 22:16:43 2011
From: skchopperguy at gmail.com (Skyler)
Date: Thu, 23 Jun 2011 13:16:43 -0700
Subject: [SR-Users] an idea - the SIP browser - elsipo
In-Reply-To: <4E0375DF.4090703@gmail.com>
References: <4E0375DF.4090703@gmail.com>
Message-ID: <008a01cc31e2$792660f0$6b7322d0$@com>

Hi Daniel,

 Great idea and thank you for releasing a working example. IMO, on the
surface this looks to me like another piece to the ultimate cross-platform
SIP communications system (elsipo + Kamailio/SERx). This project is
something I will definitely be looking at more to see how I can contribute.

 
Skyler

-----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: Thursday, June 23, 2011 10:21 AM
To: kamailio
Subject: [SR-Users] an idea - the SIP browser - elsipo

Hello,

I want to present an idea and see if it is something people find 
benefits with such approach.

Some time ago, due to existence of only SIP client applications (aka 
softphones) that have static GUI, which were designed for generic 
purposes, do not present what servces the currently used provider offers 
nor are easy to customize, I played with the idea of a SIP application 
with the GUI in HTML/CSS/Javascript.

Being busy with Kamailio, the whole thing was postponed, but during last 
days I took the time and updated the code to latest developing tools 
(QT/Webkit and pjsip) and made it public on github, see:
   * http://asipto.com/u/3c

A screenshot and some details in the wiki at:
   * http://asipto.com/u/3b

The UI can be pulled from a remote site via http/https for example. The 
approach was to have a framework to render the UI, used QT/Webkit for 
that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to 
Javascript. I worked with pjsip, but the system is plugin based, so 
pjsip plugin can be replaced, plugins for other things (not only 
sip/voip) can be written easily.

The code is just for demo/prototyping, with the goal of showing 
feasibility. At this moment can be used to make/receive calls and 
send/receive instant messaging. In addition, content of UI can be 
carried by SIP messages - the examples added so far is to request some 
server statistics when clicking on a link of UI and the SIP server 
pushing periodically some info to a section of UI.

Personally I don't have any plan to go into client side development, 
still the idea was too appealing to try and I thought would give lot of 
value and enable new features on SIP services to make user experience 
more pleasant.

Therefore the target is to see if any of you think is something that 
worth pursuing and if yes, get people on board to develop it further 
from prototype phase.

In the wiki I listed a part of my wish list with current VoIP/SIP 
services that would be trivial to implement with this approach. The 
naming as SIP browser instead of softphone hopefully suggest also the 
purpose of the application - to be able to browse the services offered 
by provider at that time and the provider to be able to push content 
back to application (on demand or periodically). Development was done on 
Mac OS X, a tarball with binaries for this OS is available for download 
on github - compiling for other OSes should go pretty straightforward.

At the moment the idea poped up there was no such approach that I could 
find (at least not in open source space), did I miss something, are you 
aware of any similar approach? What is your opinion overall?

Thanks,
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



From abalashov at evaristesys.com  Thu Jun 23 23:00:19 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 23 Jun 2011 17:00:19 -0400
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>	<4DFB8D4F.5070506@evaristesys.com>
	<BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>
Message-ID: <4E03A963.60501@evaristesys.com>

On 06/23/2011 12:58 PM, dotnetdub wrote:
>
>
> On 17 June 2011 18:22, Alex Balashov <abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>> wrote:
>
>     Brian,
>
>     Remote-Party-ID is a long-obsolete draft that never became an RFC;
>     why are you still using it?
>
>     Anyway, if you know the 'privacy' header parameter will always be
>     there, you can do this, assuming
>     is_present_hf("Remote-Party-__ID"), of course.
>
>        $var(x) = $(hdr(Remote-Party-ID){s.__select,1,>});
>
>        xlog("L_INFO", "Remote Party ID privacy value:
>     $(var(x){param.value,privacy})__\n");
>
>
>
> Hi Alex,
>
> This doesn't seem to be working for me..
>
> /sbin/kamailio[25291]: INFO: <script>: Remote Party ID privacy value:
>
>
>  From the initial invite: Remote-Party-ID: <sip:1234567 at domain.ie
> <mailto:sip%3A1234567 at domain.ie>;privacy=full;screen=no>
>
> Let me know if I'm doing something wrong.

Oh, 'privacy' is a URI parameter, not a header parameter.  (it's 
inside the angle-brackets).  That calls for a different approach:

   $var(privacy) = $(hdr(Remote-Party_ID){uri.param,privacy});

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From dotnetdub at gmail.com  Thu Jun 23 23:37:51 2011
From: dotnetdub at gmail.com (dotnetdub)
Date: Thu, 23 Jun 2011 22:37:51 +0100
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <4E03A963.60501@evaristesys.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
	<4DFB8D4F.5070506@evaristesys.com>
	<BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>
	<4E03A963.60501@evaristesys.com>
Message-ID: <BANLkTike7yFCte9mSTz_=1k6kT-hX6CimA@mail.gmail.com>

Hi Alex,

On 23 June 2011 22:00, Alex Balashov <abalashov at evaristesys.com> wrote:

> $(hdr(Remote-Party_ID){uri.**param,privacy});
>

Still not working..

ERROR: <core> [lvalue.c:351]: non existing right pvar
assignment failed at pos: (110,13-110,54)
INFO: <script>: Remote Party ID privacy value:

Thanks,
Brian
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From abalashov at evaristesys.com  Thu Jun 23 23:39:10 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 23 Jun 2011 17:39:10 -0400
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <BANLkTike7yFCte9mSTz_=1k6kT-hX6CimA@mail.gmail.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>	<4DFB8D4F.5070506@evaristesys.com>	<BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>	<4E03A963.60501@evaristesys.com>
	<BANLkTike7yFCte9mSTz_=1k6kT-hX6CimA@mail.gmail.com>
Message-ID: <4E03B27E.9060101@evaristesys.com>

On 06/23/2011 05:37 PM, dotnetdub wrote:
> Hi Alex,
>
> On 23 June 2011 22:00, Alex Balashov <abalashov at evaristesys.com
> <mailto:abalashov at evaristesys.com>> wrote:
>
>     $(hdr(Remote-Party_ID){uri.__param,privacy});
>
>
> Still not working..
>
> ERROR: <core> [lvalue.c:351]: non existing right pvar
> assignment failed at pos: (110,13-110,54)
> INFO: <script>: Remote Party ID privacy value:

Yeah, there's a typo there on my part.  Remote-Party-ID instead of 
Remote-Party_ID.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From dotnetdub at gmail.com  Thu Jun 23 23:52:50 2011
From: dotnetdub at gmail.com (dotnetdub)
Date: Thu, 23 Jun 2011 22:52:50 +0100
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <4E03B27E.9060101@evaristesys.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
	<4DFB8D4F.5070506@evaristesys.com>
	<BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>
	<4E03A963.60501@evaristesys.com>
	<BANLkTike7yFCte9mSTz_=1k6kT-hX6CimA@mail.gmail.com>
	<4E03B27E.9060101@evaristesys.com>
Message-ID: <BANLkTimwBGwTFQDJ1OkzfBM+=sqAr3sSJA@mail.gmail.com>

On 23 June 2011 22:39, Alex Balashov <abalashov at evaristesys.com> wrote:

> On 06/23/2011 05:37 PM, dotnetdub wrote:
>
>> Hi Alex,
>>
>> On 23 June 2011 22:00, Alex Balashov <abalashov at evaristesys.com
>> <mailto:abalashov at evaristesys.**com <abalashov at evaristesys.com>>> wrote:
>>
>>    $(hdr(Remote-Party_ID){uri.__**param,privacy});
>>
>>
>> Still not working..
>>
>> ERROR: <core> [lvalue.c:351]: non existing right pvar
>> assignment failed at pos: (110,13-110,54)
>> INFO: <script>: Remote Party ID privacy value:
>>
>
> Yeah, there's a typo there on my part.  Remote-Party-ID instead of
> Remote-Party_ID.
>
>
Thanks.

Just tried again.

  $var(x) = $(hdr(Remote-Party-ID){uri.param,privacy});
                xlog("L_INFO", "Remote Party ID privacy value:
$(var(x){param.value,privacy})\n");


ERROR: pv [pv_trans.c:522]: invalid uri [<sip:1234567 at domain.ie
;privacy=full;screen=no>]
/sbin/kamailio[27389]: ERROR: <core> [lvalue.c:351]: non existing right pvar
/sbin/kamailio[27389]: ERROR: <core> [lvalue.c:411]: assignment failed at
pos: (110,13-110,54)
/sbin/kamailio[27389]: INFO: <script>: Remote Party ID privacy value:






>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.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  Thu Jun 23 23:55:34 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 23 Jun 2011 17:55:34 -0400
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <BANLkTimwBGwTFQDJ1OkzfBM+=sqAr3sSJA@mail.gmail.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>	<4DFB8D4F.5070506@evaristesys.com>	<BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>	<4E03A963.60501@evaristesys.com>	<BANLkTike7yFCte9mSTz_=1k6kT-hX6CimA@mail.gmail.com>	<4E03B27E.9060101@evaristesys.com>
	<BANLkTimwBGwTFQDJ1OkzfBM+=sqAr3sSJA@mail.gmail.com>
Message-ID: <4E03B656.5020501@evaristesys.com>

On 06/23/2011 05:52 PM, dotnetdub wrote:

> ERROR: pv [pv_trans.c:522]: invalid uri [<sip:1234567 at domain.ie
> <mailto:sip%3A1234567 at domain.ie>;privacy=full;screen=no>]
> /sbin/kamailio[27389]: ERROR: <core> [lvalue.c:351]: non existing
> right pvar
> /sbin/kamailio[27389]: ERROR: <core> [lvalue.c:411]: assignment failed
> at pos: (110,13-110,54)
> /sbin/kamailio[27389]: INFO: <script>: Remote Party ID privacy value:

Oh, I guess uri.params operates on a bare URI, rather than parsing it 
out of a larger header.  That's kind of obvious, in retrospect, but 
for some reason wasn't intuitive to me.  Anyway, try it on just the URI:

    $var(x) = $(re{uri.param,privacy});

Second, make sure you're printing the value of $var(x).

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From dotnetdub at gmail.com  Fri Jun 24 00:07:17 2011
From: dotnetdub at gmail.com (dotnetdub)
Date: Thu, 23 Jun 2011 23:07:17 +0100
Subject: [SR-Users] Remote Party ID Header
In-Reply-To: <4E03B656.5020501@evaristesys.com>
References: <BANLkTi=1JY7AYnxnXt2fZdJSm4miRyadxw@mail.gmail.com>
	<4DFB8D4F.5070506@evaristesys.com>
	<BANLkTim0cQkLZ=9ct2CXN9AUxrO8auYEdw@mail.gmail.com>
	<4E03A963.60501@evaristesys.com>
	<BANLkTike7yFCte9mSTz_=1k6kT-hX6CimA@mail.gmail.com>
	<4E03B27E.9060101@evaristesys.com>
	<BANLkTimwBGwTFQDJ1OkzfBM+=sqAr3sSJA@mail.gmail.com>
	<4E03B656.5020501@evaristesys.com>
Message-ID: <BANLkTimgwg9SQH5JQhdcUhTibHDw3TT5Xg@mail.gmail.com>

On 23 June 2011 22:55, Alex Balashov <abalashov at evaristesys.com> wrote:

> On 06/23/2011 05:52 PM, dotnetdub wrote:
>
>  ERROR: pv [pv_trans.c:522]: invalid uri [<sip:1234567 at domain.ie
>> <mailto:sip%3A1234567 at domain.**ie <sip%253A1234567 at domain.ie>
>> >;privacy=full;screen=no>]
>>
>> /sbin/kamailio[27389]: ERROR: <core> [lvalue.c:351]: non existing
>> right pvar
>> /sbin/kamailio[27389]: ERROR: <core> [lvalue.c:411]: assignment failed
>> at pos: (110,13-110,54)
>> /sbin/kamailio[27389]: INFO: <script>: Remote Party ID privacy value:
>>
>
> Oh, I guess uri.params operates on a bare URI, rather than parsing it out
> of a larger header.  That's kind of obvious, in retrospect, but for some
> reason wasn't intuitive to me.  Anyway, try it on just the URI:
>
>   $var(x) = $(re{uri.param,privacy});
>
> Second, make sure you're printing the value of $var(x).



Thanks

Works perfectly.



>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.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 miconda at gmail.com  Fri Jun 24 09:09:31 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Jun 2011 09:09:31 +0200
Subject: [SR-Users] Don't publish Is not supposed the e-mail should be
 Hiden
In-Reply-To: <E30A9B5D-27C5-45A6-A265-503064EA9D93@321communications.com>
References: <E30A9B5D-27C5-45A6-A265-503064EA9D93@321communications.com>
Message-ID: <4E04382B.6080606@gmail.com>



On 6/23/11 3:05 AM, Omar wrote:
> I really don't want to share my e-mail publicly, i remember there was an option before.
Is it about a mailman list manager option? There was nothing changed in 
this regard. Can you point to the documentation of this option?

Thanks,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From zearush at 163.com  Fri Jun 24 03:58:32 2011
From: zearush at 163.com (=?GBK?B?zuLD9w==?=)
Date: Fri, 24 Jun 2011 09:58:32 +0800 (CST)
Subject: [SR-Users] Hello,can you send me the useage of siremis?Thank you
Message-ID: <35c6b5f7.12774.130bf5eb2c6.Coremail.zearush@163.com>

HI:
    I'm a user of siremis from China.Now I install the siremis successfully.But I would not use it. So I want the useage of it.
    I'll wait for your letter.Thank you!


Best regards.
Calvin wu
2011-6-24
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From miconda at gmail.com  Fri Jun 24 09:58:25 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Jun 2011 09:58:25 +0200
Subject: [SR-Users] MYSQL Compilation Kamailio 3.1.4
In-Reply-To: <21FAF94F-ADCB-4714-8154-3D43E1308D0D@321communications.com>
References: <21FAF94F-ADCB-4714-8154-3D43E1308D0D@321communications.com>
Message-ID: <4E0443A1.4070509@gmail.com>

Can you try adding

#include <netinet/in.h>

before first #include in modules/db_mysql/my_cmd.c file?

If not working, try to grep in /usr/include/ to see which is the file 
where sin6 is defined.

Cheers,
Daniel

On 6/23/11 3:03 AM, Omar wrote:
> bash-3.00# make clean
> bash-3.00# make quiet=0
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c km_db_mysql.c -o km_db_mysql.o -MMD -MP
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_uri.c -o my_uri.o -MMD -MP
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_res.c -o my_res.o -MMD -MP
> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_cmd.c -o my_cmd.o -MMD -MP
> In file included from ../../globals.h:37,
>                   from ../../pt.h:42,
>                   from ../../counters.h:43,
>                   from mysql_mod.h:39,
>                   from my_cmd.c:40:
> ../../ip_addr.h:86: error: field 'sin6' has incomplete type
> ../../ip_addr.h: In function 'ip_addr_loopback':
> ../../ip_addr.h:283: warning: implicit declaration of function 'htonl'
> ../../ip_addr.h:286: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK'
> ../../ip_addr.h: In function 'sockaddr2ip_addr':
> ../../ip_addr.h:353: error: dereferencing pointer to incomplete type
> ../../ip_addr.h: In function 'su_getport':
> ../../ip_addr.h:399: warning: implicit declaration of function 'ntohs'
> ../../ip_addr.h: In function 'su_setport':
> ../../ip_addr.h:418: warning: implicit declaration of function 'htons'
> make: *** [my_cmd.o] Error 1
>
>
> the onyl thing i did is uncomment the CROSSCOMPILE line
>
> regards
>
> 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

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda




From vaad.fabi at gmail.com  Fri Jun 24 10:01:11 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Fri, 24 Jun 2011 11:01:11 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E008EFC.3020001@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
	<4E008EFC.3020001@gmail.com>
Message-ID: <4E044447.6030304@gmail.com>

  Hi all,

Still trying to run kamailio 3.1.x + asterisk 1.6.2.X realtime architecture.
As i mentioned earlier, instead of asterisk i use another softswitch 
with db, table "retailroute" with "login" and "password" fields as user 
auth data.
Of course, troubles... user can't register to kamailio :(

Logs( without users attempts to register, only kamailio internal logic:


Jun 24 11:23:26 sbc kamailio[17768]: INFO: <script>: ------------------ 
SIP Request: method [NOTIFY] from [sip:100 at 100.100.100.100] to 
[sip:100.100.100.100]
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
[parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <811> <NOTIFY>
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: maxfwd [mf_funcs.c:85]: 
value = 70
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
[parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
[parser/msg_parser.c:103]: found end of header
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
[msg_translator.c:204]: check_via_address(XX.116.255.XXX, 192.168.0.180, 0)
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: siputils [checks.c:73]: no 
totag
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1081]: DEBUG: 
t_check_msg: msg id=39 global id=38 T start=(nil)
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:528]: 
t_lookup_request: start searching: hash=31430, isACK=0
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:485]: DEBUG: 
RFC3261 transaction matching failed
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:711]: DEBUG: 
t_lookup_request: no transaction found
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1150]: DEBUG: 
t_check_msg: msg id=39 global id=39 T end=(nil)
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [select.c:424]: 
Calling SELECT 0x832b7e4
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [socket_info.c:501]: 
grep_sock_info - checking if host==us: 13==13 &&  [100.100.100.100] == 
[100.100.100.100]
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [socket_info.c:504]: 
grep_sock_info - checking if port 5060 matches port 5060
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db [authorize.c:239]: 
realm value [100.100.100.100]
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:95]: auth: 
digest-algo: MD5 parsed value: 1
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:118]: 
allocate 28 bytes for result set at 0x832c8cc
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:68]: 1 
columns returned from the query
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:155]: 
allocate 4 bytes for result names at 0x832ba40
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:165]: 
allocate 4 bytes for result types at 0x832ba08
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:85]: 
allocate 8 bytes for RES_NAMES[0] at 0x832c950
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:92]: 
RES_NAMES(0x832c950)[0]=[password]
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:136]: use 
DB1_STRING result type
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:184]: 
allocate 8 bytes for rows at 0x832c918
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:119]: 
allocate 20 bytes for row values at 0x832e75c
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_val.c:117]: 
converting STRING [dav0011]
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db [authorize.c:120]: 
HA1 string calculated: a8fdf502e497ff8e9407ac2dc31540df
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:205]: 
check_response: Our result = 'fa83bf7c0a7af33cc500aa2b62758931'
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:215]: 
check_response: Authorization failed
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:81]: 
freeing 1 columns
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:85]: 
freeing RES_NAMES[0] at 0x832c950
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:94]: 
freeing result names at 0x832ba40
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:99]: 
freeing result types at 0x832ba08
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:54]: 
freeing 1 rows
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:97]: 
freeing row values at 0x832e75c
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:62]: 
freeing rows at 0x832c918
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:136]: 
freeing result set at 0x832c8cc
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:102]: 
build_challenge_hf: realm='100.100.100.100'
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:236]: 
auth: 'Proxy-Authenticate: Digest realm="100.100.100.100", 
nonce="TgQ8mU4EO22Odd1W63tOO27IRrKX0kU '
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: sl [sl.c:278]: reply in 
stateless mode (sl)
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
[msg_translator.c:204]: check_via_address(XX.116.255.XXX, 192.168.0.180, 0)
Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [usr_avp.c:646]: 
DEBUG:destroy_avp_list: destroying list (nil)



Such messages periodically sends kamailio. Why kamailio every time sends 
yourself these NOTIFY messages, what it try to do ?
I see problem something around auth_db autorization and HA1, isn't it ? 
How to fix it ?


-- 
Vadim F.



From henning.westerholt at 1und1.de  Fri Jun 24 10:29:52 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Fri, 24 Jun 2011 10:29:52 +0200
Subject: [SR-Users] Don't publish Is not supposed the e-mail should be
	Hiden
In-Reply-To: <4E04382B.6080606@gmail.com>
References: <E30A9B5D-27C5-45A6-A265-503064EA9D93@321communications.com>
	<4E04382B.6080606@gmail.com>
Message-ID: <201106241029.52859.henning.westerholt@1und1.de>

On Friday 24 June 2011, Daniel-Constantin Mierla wrote:
> > I really don't want to share my e-mail publicly, i remember there was an
> > option before.
> 
> Is it about a mailman list manager option? There was nothing changed in
> this regard. Can you point to the documentation of this option?

Hi Omar,

this list is configured in a way that normal subscribers can't see the list of 
the other members. Also the archives obscure the shown mail addresses. But if 
you want to participate in the discussions, you obviously need a mail address, 
which everybody else subscribed can see as well.

Best regards,

Henning


From miconda at gmail.com  Fri Jun 24 11:00:33 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Jun 2011 11:00:33 +0200
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E044447.6030304@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
	<4E008EFC.3020001@gmail.com> <4E044447.6030304@gmail.com>
Message-ID: <4E045231.9080906@gmail.com>

Hello,

based on the logs, the authentication fails due to wrong password. Check 
that you have the right password on client side.

Kamailio does not generate any NOTIFY unless the presence server modules 
are loaded. If not, then the notify comes from somewhere else and it is 
just looped back due to config file routing logic.

Cheers,
Daniel

On 6/24/11 10:01 AM, vaad.fabi at gmail.com wrote:
>  Hi all,
>
> Still trying to run kamailio 3.1.x + asterisk 1.6.2.X realtime 
> architecture.
> As i mentioned earlier, instead of asterisk i use another softswitch 
> with db, table "retailroute" with "login" and "password" fields as 
> user auth data.
> Of course, troubles... user can't register to kamailio :(
>
> Logs( without users attempts to register, only kamailio internal logic:
>
>
> Jun 24 11:23:26 sbc kamailio[17768]: INFO: <script>: 
> ------------------ SIP Request: method [NOTIFY] from 
> [sip:100 at 100.100.100.100] to [sip:100.100.100.100]
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <811> <NOTIFY>
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: maxfwd [mf_funcs.c:85]: 
> value = 70
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [parser/msg_parser.c:103]: found end of header
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [msg_translator.c:204]: check_via_address(XX.116.255.XXX, 
> 192.168.0.180, 0)
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: siputils [checks.c:73]: no 
> totag
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1081]: 
> DEBUG: t_check_msg: msg id=39 global id=38 T start=(nil)
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:528]: 
> t_lookup_request: start searching: hash=31430, isACK=0
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:485]: 
> DEBUG: RFC3261 transaction matching failed
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:711]: 
> DEBUG: t_lookup_request: no transaction found
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1150]: 
> DEBUG: t_check_msg: msg id=39 global id=39 T end=(nil)
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [select.c:424]: 
> Calling SELECT 0x832b7e4
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==13 &&  
> [100.100.100.100] == [100.100.100.100]
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches 
> port 5060
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db [authorize.c:239]: 
> realm value [100.100.100.100]
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:95]: auth: 
> digest-algo: MD5 parsed value: 1
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:118]: 
> allocate 28 bytes for result set at 0x832c8cc
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:68]: 1 
> columns returned from the query
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:155]: 
> allocate 4 bytes for result names at 0x832ba40
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:165]: 
> allocate 4 bytes for result types at 0x832ba08
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:85]: 
> allocate 8 bytes for RES_NAMES[0] at 0x832c950
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:92]: 
> RES_NAMES(0x832c950)[0]=[password]
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:136]: 
> use DB1_STRING result type
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:184]: 
> allocate 8 bytes for rows at 0x832c918
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:119]: 
> allocate 20 bytes for row values at 0x832e75c
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_val.c:117]: 
> converting STRING [dav0011]
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db [authorize.c:120]: 
> HA1 string calculated: a8fdf502e497ff8e9407ac2dc31540df
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:205]: 
> check_response: Our result = 'fa83bf7c0a7af33cc500aa2b62758931'
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:215]: 
> check_response: Authorization failed
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:81]: 
> freeing 1 columns
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:85]: 
> freeing RES_NAMES[0] at 0x832c950
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:94]: 
> freeing result names at 0x832ba40
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:99]: 
> freeing result types at 0x832ba08
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:54]: 
> freeing 1 rows
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:97]: 
> freeing row values at 0x832e75c
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:62]: 
> freeing rows at 0x832c918
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:136]: 
> freeing result set at 0x832c8cc
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:102]: 
> build_challenge_hf: realm='100.100.100.100'
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:236]: 
> auth: 'Proxy-Authenticate: Digest realm="100.100.100.100", 
> nonce="TgQ8mU4EO22Odd1W63tOO27IRrKX0kU '
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: sl [sl.c:278]: reply in 
> stateless mode (sl)
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
> [msg_translator.c:204]: check_via_address(XX.116.255.XXX, 
> 192.168.0.180, 0)
> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [usr_avp.c:646]: 
> DEBUG:destroy_avp_list: destroying list (nil)
>
>
>
> Such messages periodically sends kamailio. Why kamailio every time 
> sends yourself these NOTIFY messages, what it try to do ?
> I see problem something around auth_db autorization and HA1, isn't it 
> ? How to fix it ?
>
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Fri Jun 24 11:03:43 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Jun 2011 11:03:43 +0200
Subject: [SR-Users] Dispatcher module for INVITE
In-Reply-To: <BANLkTi=ToWS5FHmDx=8RK5Xpxic8-x--2w@mail.gmail.com>
References: <BANLkTi=ToWS5FHmDx=8RK5Xpxic8-x--2w@mail.gmail.com>
Message-ID: <4E0452EF.90708@gmail.com>

Hello,

if you use forward() then the retransmission is not from kamailio - 
forward() does stateless forwarding.

The dispatcher module readme in devel branch has a better example, maybe 
it helps you:
http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2513148

Cheers,
Daniel

On 6/23/11 8:35 AM, Giuseppe Carella wrote:
> Hi all,
>
> I'm using kamailio dispatcher module between
> the OpenIMSCore and sailfin. Normally, without the dispatcher,
> all works fine. From the IMS Bench I can see that there are
> many successful call, but when I use the dispatcher module
> I'm having a lot of retransmission in the first part of the INVITE
> protocol. I'm using the default configuration file:
>
> # ----------------- setting module-specific parameters ---------------
> # -- dispatcher params --
> modparam("dispatcher", "list_file", "../etc/dispatcher.list")
> # modparam("dispatcher", "force_dst", 1)
> route{
> if ( !mf_process_maxfwd_header("10") )
> {
> sl_send_reply("483","To Many Hops");
> drop();
> };
> ds_select_dst("1", "0");
> forward();
> # t_relay();
> }
>
>
> I tried also the t_relay() function instead of the forward
> command, but the problem is the same.
>
> Thanks in advance.
>
> Bests,
>
> Giuseppe
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From miconda at gmail.com  Fri Jun 24 11:17:59 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Fri, 24 Jun 2011 11:17:59 +0200
Subject: [SR-Users] an idea - the SIP browser - elsipo
In-Reply-To: <008a01cc31e2$792660f0$6b7322d0$@com>
References: <4E0375DF.4090703@gmail.com> <008a01cc31e2$792660f0$6b7322d0$@com>
Message-ID: <4E045647.5020403@gmail.com>

Hi Skyler,

On 6/23/11 10:16 PM, Skyler wrote:
> Hi Daniel,
>
>   Great idea and thank you for releasing a working example. IMO, on the
> surface this looks to me like another piece to the ultimate cross-platform
> SIP communications system (elsipo + Kamailio/SERx). This project is
> something I will definitely be looking at more to see how I can contribute.
thanks! This was the purpose - to get a communication system beyond 
classic "cold" telephony interaction. And proved to be rather simple, 
the whole UI for the example in GT right now is less that 400 lines (the 
HTML+CSS+Javascript). Moreover, the UI can be changed very easy.

So far I haven't met a decent interaction between users and providers 
(core network), while many things can be done straightforward with just 
a click. Instead of using DTMF, with pin auth and so on, why not just a 
click to:
- set DND on server and redirect to voicemail
- tell the server to email the last voice message
- block incoming call from specific numbers
- and the list can continue -- all from your SIP application so it is no 
need to login via web or what so ever, the SIP server can handle such 
requests safe (auth+tls) and very convenient in the config file, with 
our language or lua/perl/python...

If C++ is not your area, tuning the HTML/CSS/Javascript to make the UI 
more attractive can be a way to contribute.

Cheers,
Daniel


>
>
> Skyler
>
> -----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: Thursday, June 23, 2011 10:21 AM
> To: kamailio
> Subject: [SR-Users] an idea - the SIP browser - elsipo
>
> Hello,
>
> I want to present an idea and see if it is something people find
> benefits with such approach.
>
> Some time ago, due to existence of only SIP client applications (aka
> softphones) that have static GUI, which were designed for generic
> purposes, do not present what servces the currently used provider offers
> nor are easy to customize, I played with the idea of a SIP application
> with the GUI in HTML/CSS/Javascript.
>
> Being busy with Kamailio, the whole thing was postponed, but during last
> days I took the time and updated the code to latest developing tools
> (QT/Webkit and pjsip) and made it public on github, see:
>     * http://asipto.com/u/3c
>
> A screenshot and some details in the wiki at:
>     * http://asipto.com/u/3b
>
> The UI can be pulled from a remote site via http/https for example. The
> approach was to have a framework to render the UI, used QT/Webkit for
> that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to
> Javascript. I worked with pjsip, but the system is plugin based, so
> pjsip plugin can be replaced, plugins for other things (not only
> sip/voip) can be written easily.
>
> The code is just for demo/prototyping, with the goal of showing
> feasibility. At this moment can be used to make/receive calls and
> send/receive instant messaging. In addition, content of UI can be
> carried by SIP messages - the examples added so far is to request some
> server statistics when clicking on a link of UI and the SIP server
> pushing periodically some info to a section of UI.
>
> Personally I don't have any plan to go into client side development,
> still the idea was too appealing to try and I thought would give lot of
> value and enable new features on SIP services to make user experience
> more pleasant.
>
> Therefore the target is to see if any of you think is something that
> worth pursuing and if yes, get people on board to develop it further
> from prototype phase.
>
> In the wiki I listed a part of my wish list with current VoIP/SIP
> services that would be trivial to implement with this approach. The
> naming as SIP browser instead of softphone hopefully suggest also the
> purpose of the application - to be able to browse the services offered
> by provider at that time and the provider to be able to push content
> back to application (on demand or periodically). Development was done on
> Mac OS X, a tarball with binaries for this OS is available for download
> on github - compiling for other OSes should go pretty straightforward.
>
> At the moment the idea poped up there was no such approach that I could
> find (at least not in open source space), did I miss something, are you
> aware of any similar approach? What is your opinion overall?
>
> Thanks,
> Daniel
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From mgwilliams at gmail.com  Fri Jun 24 13:00:13 2011
From: mgwilliams at gmail.com (Matthew Williams)
Date: Fri, 24 Jun 2011 07:00:13 -0400
Subject: [SR-Users] async processing in a module
Message-ID: <BANLkTi=NqNeeGp+E1jRQrY_M-fNK4fGBdQ@mail.gmail.com>

Hi,

I'm working on a jsonrpc (client) module that we would like to make async,
using t_suspend and t_continue from the tm module. I have a couple questions
related to this.

The way I understand these functions, t_suspend will freeze the transaction
in shared memory and immediately return and t_continue will unfreeze the
transaction. What I'm not clear on here is whether t_continue will continue
processing of the transaction within the process that calls t_continue, or
will it delegate to a worker process?

I assume I should spawn an io process in child_init which will be
responsible for sending/receiving on the socket(s), but what is the best way
to communicate between processes? Is there a way to get a fd from
fork_process (that can be written to by any child and read from using
select() in the io thread)?

This is my first attempt at a real module, so any pointers are greatly
appreciated.

Regards,
Matthew WIlliams
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From gminghon at gmail.com  Fri Jun 24 14:01:32 2011
From: gminghon at gmail.com (MingHon)
Date: Fri, 24 Jun 2011 20:01:32 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
Message-ID: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>

hey list,


i currently test on my asterisk 1.6 box on on pc with private ip

and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc
with public ip and private ip.

everything installed successfully. i follow this tutorial realtime
intergration with asterisk

http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb


my config.

rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user


#!define WITH_NAT

modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

modparam("nathelper", "sipping_from", "sip:pinger at publicip")

nat_uac_test("19")

# uncomment next line to do SIP NAT pinging

setbflag(FLB_NATSIPPING);

is my config correct?

may i know the best version work out from the box?

please advice..

thanks in adv. :)

-- 
Regards,

MingHon
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From carsten at ng-voice.com  Fri Jun 24 16:29:15 2011
From: carsten at ng-voice.com (Carsten Bock)
Date: Fri, 24 Jun 2011 16:29:15 +0200
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTin-vcyPEANFRb-OKwxcravmLv-Lqw@mail.gmail.com>

Hi,

did you also set

#!define WITH_NAT

at the beginning of your config?
That should do the job.

Carsten

2011/6/24 MingHon <gminghon at gmail.com>:
> hey list,
>
> i currently test on my asterisk 1.6 box on on pc with private ip
> and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc
> with public ip and private ip.
> everything installed successfully. i follow this tutorial realtime
> intergration with asterisk
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
>
> my config.
> rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user
>
> #!define WITH_NAT
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
> modparam("nathelper", "sipping_from", "sip:pinger at publicip")
> nat_uac_test("19")
> # uncomment next line to do SIP NAT pinging
> setbflag(FLB_NATSIPPING);
> is my config correct?
> may i know the best version work out from the box?
> please advice..
> thanks in adv. :)
> --
> Regards,
>
> MingHon
>
> _______________________________________________
> 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
http://www.ng-voice.com
mailto:carsten at ng-voice.com

Schomburgstr. 80
22767 Hamburg
Germany

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220


From gminghon at gmail.com  Fri Jun 24 17:55:53 2011
From: gminghon at gmail.com (MingHon)
Date: Fri, 24 Jun 2011 23:55:53 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTim4O_SK7KfzyLsTsBNWifegfYSpAw@mail.gmail.com>

Hi,

Yup i did define #!define WITH_NAT at the beginning of config file. it
doesnt work.

been struggle for a month hope you can help.

and also in the cfg im listening to both iface.


listen=public ip

listen=192.168.2.3 [kamailio ip]


Regards,

MingHon
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From stefano.larosa at 10993.it  Fri Jun 24 19:01:55 2011
From: stefano.larosa at 10993.it (Stefano Larosa)
Date: Fri, 24 Jun 2011 19:01:55 +0200
Subject: [SR-Users] R: rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTim4O_SK7KfzyLsTsBNWifegfYSpAw@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
	<BANLkTim4O_SK7KfzyLsTsBNWifegfYSpAw@mail.gmail.com>
Message-ID: <01bc01cc3290$6bb5fcb0$4321f610$@larosa@10993.it>

Hi,

 

i should check your firewall settings.

 

Regards,

Stefano Larosa.

 

Da: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] Per conto di MingHon
Inviato: venerd? 24 giugno 2011 17.56
A: sr-users at lists.sip-router.org
Oggetto: Re: [SR-Users] rtpproxy and kamailio doesnt work out from the box.

 

Hi,

 

Yup i did define #!define WITH_NAT at the beginning of config file. it
doesnt work.

 

been struggle for a month hope you can help.

 

and also in the cfg im listening to both iface.

 





listen=public ip





listen=192.168.2.3 [kamailio ip]





 

Regards,

MingHon

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From vaad.fabi at gmail.com  Fri Jun 24 20:12:37 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Fri, 24 Jun 2011 21:12:37 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E045231.9080906@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
	<4E008EFC.3020001@gmail.com> <4E044447.6030304@gmail.com>
	<4E045231.9080906@gmail.com>
Message-ID: <4E04D395.4020709@gmail.com>

  Yes, NOTIFY was generated by ipphone.

Thx, again :)


On 06/24/2011 12:00 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> based on the logs, the authentication fails due to wrong password. 
> Check that you have the right password on client side.
>
> Kamailio does not generate any NOTIFY unless the presence server 
> modules are loaded. If not, then the notify comes from somewhere else 
> and it is just looped back due to config file routing logic.
>
> Cheers,
> Daniel
>
> On 6/24/11 10:01 AM, vaad.fabi at gmail.com wrote:
>>  Hi all,
>>
>> Still trying to run kamailio 3.1.x + asterisk 1.6.2.X realtime 
>> architecture.
>> As i mentioned earlier, instead of asterisk i use another softswitch 
>> with db, table "retailroute" with "login" and "password" fields as 
>> user auth data.
>> Of course, troubles... user can't register to kamailio :(
>>
>> Logs( without users attempts to register, only kamailio internal logic:
>>
>>
>> Jun 24 11:23:26 sbc kamailio[17768]: INFO: <script>: 
>> ------------------ SIP Request: method [NOTIFY] from 
>> [sip:100 at 100.100.100.100] to [sip:100.100.100.100]
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <811> <NOTIFY>
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: maxfwd [mf_funcs.c:85]: 
>> value = 70
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [parser/msg_parser.c:103]: found end of header
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [msg_translator.c:204]: check_via_address(XX.116.255.XXX, 
>> 192.168.0.180, 0)
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: siputils [checks.c:73]: 
>> no totag
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1081]: 
>> DEBUG: t_check_msg: msg id=39 global id=38 T start=(nil)
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:528]: 
>> t_lookup_request: start searching: hash=31430, isACK=0
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:485]: 
>> DEBUG: RFC3261 transaction matching failed
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:711]: 
>> DEBUG: t_lookup_request: no transaction found
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1150]: 
>> DEBUG: t_check_msg: msg id=39 global id=39 T end=(nil)
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [select.c:424]: 
>> Calling SELECT 0x832b7e4
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==13 
>> &&  [100.100.100.100] == [100.100.100.100]
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches 
>> port 5060
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db 
>> [authorize.c:239]: realm value [100.100.100.100]
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:95]: auth: 
>> digest-algo: MD5 parsed value: 1
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:118]: 
>> allocate 28 bytes for result set at 0x832c8cc
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:68]: 1 
>> columns returned from the query
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:155]: 
>> allocate 4 bytes for result names at 0x832ba40
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:165]: 
>> allocate 4 bytes for result types at 0x832ba08
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:85]: 
>> allocate 8 bytes for RES_NAMES[0] at 0x832c950
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:92]: 
>> RES_NAMES(0x832c950)[0]=[password]
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:136]: 
>> use DB1_STRING result type
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:184]: 
>> allocate 8 bytes for rows at 0x832c918
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:119]: 
>> allocate 20 bytes for row values at 0x832e75c
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_val.c:117]: 
>> converting STRING [dav0011]
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db 
>> [authorize.c:120]: HA1 string calculated: 
>> a8fdf502e497ff8e9407ac2dc31540df
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:205]: 
>> check_response: Our result = 'fa83bf7c0a7af33cc500aa2b62758931'
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:215]: 
>> check_response: Authorization failed
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:81]: 
>> freeing 1 columns
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:85]: 
>> freeing RES_NAMES[0] at 0x832c950
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:94]: 
>> freeing result names at 0x832ba40
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:99]: 
>> freeing result types at 0x832ba08
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:54]: 
>> freeing 1 rows
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:97]: 
>> freeing row values at 0x832e75c
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:62]: 
>> freeing rows at 0x832c918
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:136]: 
>> freeing result set at 0x832c8cc
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:102]: 
>> build_challenge_hf: realm='100.100.100.100'
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:236]: 
>> auth: 'Proxy-Authenticate: Digest realm="100.100.100.100", 
>> nonce="TgQ8mU4EO22Odd1W63tOO27IRrKX0kU '
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: sl [sl.c:278]: reply in 
>> stateless mode (sl)
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>> [msg_translator.c:204]: check_via_address(XX.116.255.XXX, 
>> 192.168.0.180, 0)
>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [usr_avp.c:646]: 
>> DEBUG:destroy_avp_list: destroying list (nil)
>>
>>
>>
>> Such messages periodically sends kamailio. Why kamailio every time 
>> sends yourself these NOTIFY messages, what it try to do ?
>> I see problem something around auth_db autorization and HA1, isn't it 
>> ? How to fix it ?
>>
>>
>


-- 
vad fabi



From fborot at hotmail.com  Fri Jun 24 22:28:28 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Fri, 24 Jun 2011 16:28:28 -0400
Subject: [SR-Users] kamailio fails to start with kamctl start
Message-ID: <SNT110-W64C60382EE6B2F2796781CD520@phx.gbl>


Hello
I just installed kamailio-3.1 and using the default config file "kamctl start" fails


 [root at registrar sbin]# ./kamctl start

INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
[root at registrar sbin]# 

but when I use this command it starts: <kamailio -f path_to_config_file

[root at registrar sbin]# ./kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg
loading modules under /usr/local/kamailio-3.1//lib64/kamailio/modules_k/:/usr/local/kamailio-3.1//lib64/kamailio/modules/
Listening on
             udp: 127.0.0.1:5060
             udp: 192.168.168.10:5060
             tcp: 127.0.0.1:5060
             tcp: 192.168.168.10:5060
Aliases:
             tcp: localhost:5060
             tcp: localhost.localdomain:5060
             tcp: registrar:5060
             tcp: registrar.mytestdomain.com:5060
             udp: localhost:5060
             udp: localhost.localdomain:5060
             udp: registrar:5060
             udp: registrar.mytestdomain.com:5060

[root at registrar sbin]#

Also, when I use "./kamctl start" the syslog file does not show anything at all that would help me why it is failing. 
But when I use "/usr/local/kamailio-3.1/sbin/kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg" it does update the syslog file.

Also, after kamailio start with "./kamailio -f .." I can use ./kamctl fifo <any option>. Example:

[root at registrar sbin]# ./kamctl fifo version
Server:: kamailio (3.1.4 (x86_64/linux))
Build:: mi_core.c compiled on 15:09:40 Jun 24 2011 with gcc 4.1.2
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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
GIT:: e2d6f0
[root at registrar sbin]#

[root at registrar sbin]# ./kamctl fifo ps
Process::  ID=0 PID=27078 Type=attendant
Process::  ID=1 PID=27079 Type=udp receiver child=0 sock=127.0.0.1:5060
Process::  ID=2 PID=27080 Type=udp receiver child=1 sock=127.0.0.1:5060
Process::  ID=3 PID=27081 Type=udp receiver child=2 sock=127.0.0.1:5060
Process::  ID=4 PID=27082 Type=udp receiver child=3 sock=127.0.0.1:5060
Process::  ID=5 PID=27083 Type=udp receiver child=0 sock=192.168.168.10:5060
Process::  ID=6 PID=27084 Type=udp receiver child=1 sock=192.168.168.10:5060
Process::  ID=7 PID=27085 Type=udp receiver child=2 sock=192.168.168.10:5060
Process::  ID=8 PID=27086 Type=udp receiver child=3 sock=192.168.168.10:5060
Process::  ID=9 PID=27087 Type=slow timer
Process::  ID=10 PID=27088 Type=timer
Process::  ID=11 PID=27089 Type=MI FIFO
Process::  ID=12 PID=27090 Type=ctl handler
Process::  ID=13 PID=27091 Type=tcp receiver child=0
Process::  ID=14 PID=27092 Type=tcp receiver child=1
Process::  ID=15 PID=27093 Type=tcp receiver child=2
Process::  ID=16 PID=27094 Type=tcp receiver child=3
Process::  ID=17 PID=27095 Type=tcp main process
[root at registrar sbin]#



 		 	   		  
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From fborot at hotmail.com  Sat Jun 25 00:10:38 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Fri, 24 Jun 2011 18:10:38 -0400
Subject: [SR-Users] kamailio fails to start with kamctl start
In-Reply-To: <SNT110-W64C60382EE6B2F2796781CD520@phx.gbl>
References: <SNT110-W64C60382EE6B2F2796781CD520@phx.gbl>
Message-ID: <SNT110-W4593A88C65D2250B77EE35CD520@phx.gbl>



I enabled debugging on the kamctl script and see these lines right before it fails:

+ minfo 'Starting Kamailio : '
+ '[' -t 1 -a -z '' ']'
+ echo -e '\E[37;33m\033[1mINFO: Starting Kamailio : \033[0m'
INFO: Starting Kamailio :
+ '[' -r /var/run/kamailio.pid ']'
+ '[' '!' -x ./kamailio ']'
+ '[' 1 = 1 ']'
+ ./kamailio -P /var/run/kamailio.pid
+ sleep 3
+ '[' '!' -s /var/run/kamailio.pid ']'
+ echo

+ merr 'PID file /var/run/kamailio.pid does not exist -- Kamailio start failed'
+ '[' -t 1 -a -z '' ']'
+ echo -e '\E[37;31m\033[1mERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed\033[0m'
ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
+ exit 1
[root at registrar sbin]#


these are the main steps I followed to install it:

mkdir -p /usr/local/src/kamailio-3.1.0
cd /usr/local/src/kamailio-3.1.0

git clone --depth 1 git://git.sip-router.org/sip-router kamailio
cd kamailio
git checkout -b 3.1 origin/3.1

mkdir /usr/local/kamailio-3.1

make FLAVOUR=kamailio cfg

vi modules.lst to exclude db_mysql db_odbc from modules.lst 

make PREFIX="/usr/local/kamailio-3.1/" all
make PREFIX="/usr/local/kamailio-3.1/" install


then I executed /usr/local/kamailio-3.1/sbin/kamdbctl create to create the DB and tables.



From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Date: Fri, 24 Jun 2011 16:28:28 -0400
Subject: [SR-Users] kamailio fails to start with kamctl start








Hello
I just installed kamailio-3.1 and using the default config file "kamctl start" fails


 [root at registrar sbin]# ./kamctl start

INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
[root at registrar sbin]# 

but when I use this command it starts: <kamailio -f path_to_config_file

[root at registrar sbin]# ./kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg
loading modules under /usr/local/kamailio-3.1//lib64/kamailio/modules_k/:/usr/local/kamailio-3.1//lib64/kamailio/modules/
Listening on
             udp: 127.0.0.1:5060
             udp: 192.168.168.10:5060
             tcp: 127.0.0.1:5060
             tcp: 192.168.168.10:5060
Aliases:
             tcp: localhost:5060
             tcp: localhost.localdomain:5060
             tcp: registrar:5060
             tcp: registrar.mytestdomain.com:5060
             udp: localhost:5060
             udp: localhost.localdomain:5060
             udp: registrar:5060
             udp: registrar.mytestdomain.com:5060

[root at registrar sbin]#

Also, when I use "./kamctl start" the syslog file does not show anything at all that would help me why it is failing. 
But when I use "/usr/local/kamailio-3.1/sbin/kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg" it does update the syslog file.

Also, after kamailio start with "./kamailio -f .." I can use ./kamctl fifo <any option>. Example:

[root at registrar sbin]# ./kamctl fifo version
Server:: kamailio (3.1.4 (x86_64/linux))
Build:: mi_core.c compiled on 15:09:40 Jun 24 2011 with gcc 4.1.2
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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
GIT:: e2d6f0
[root at registrar sbin]#

[root at registrar sbin]# ./kamctl fifo ps
Process::  ID=0 PID=27078 Type=attendant
Process::  ID=1 PID=27079 Type=udp receiver child=0 sock=127.0.0.1:5060
Process::  ID=2 PID=27080 Type=udp receiver child=1 sock=127.0.0.1:5060
Process::  ID=3 PID=27081 Type=udp receiver child=2 sock=127.0.0.1:5060
Process::  ID=4 PID=27082 Type=udp receiver child=3 sock=127.0.0.1:5060
Process::  ID=5 PID=27083 Type=udp receiver child=0 sock=192.168.168.10:5060
Process::  ID=6 PID=27084 Type=udp receiver child=1 sock=192.168.168.10:5060
Process::  ID=7 PID=27085 Type=udp receiver child=2 sock=192.168.168.10:5060
Process::  ID=8 PID=27086 Type=udp receiver child=3 sock=192.168.168.10:5060
Process::  ID=9 PID=27087 Type=slow timer
Process::  ID=10 PID=27088 Type=timer
Process::  ID=11 PID=27089 Type=MI FIFO
Process::  ID=12 PID=27090 Type=ctl handler
Process::  ID=13 PID=27091 Type=tcp receiver child=0
Process::  ID=14 PID=27092 Type=tcp receiver child=1
Process::  ID=15 PID=27093 Type=tcp receiver child=2
Process::  ID=16 PID=27094 Type=tcp receiver child=3
Process::  ID=17 PID=27095 Type=tcp main process
[root at registrar sbin]#



 		 	   		  

_______________________________________________
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 gminghon at gmail.com  Sat Jun 25 05:28:01 2011
From: gminghon at gmail.com (MingHon)
Date: Sat, 25 Jun 2011 11:28:01 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTi=UfrQPHMBSBRYTgYhsG1GNkAWJ1g@mail.gmail.com>

Hi,

which firewall do you mean? the uac firewall or the kamailio firewall?

kamailio doesnt have firewall. kamailio and rtpproxy in centos 5.4 firewall
and selinux disabled.

eth0 is configured for ppp0e and eth1 is the private ip and

echo "1" > /proc/sys/net/ipv4/ip_forward

iptables -t nat -A POSTROUTING -o ppp0 -j MASQUERADE

and for asterisk is in fedora14 firewall and selinux also disabled.

for uac the firewall also disabled in the router. (dlink dir-615).

what else do i need to check?  pls adv..


-- 
Regards,

MingHon
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From roman at dmytriv.com  Mon Jun 27 01:24:03 2011
From: roman at dmytriv.com (Roman Dmytriv)
Date: Sun, 26 Jun 2011 19:24:03 -0400
Subject: [SR-Users] Kamailio start failed
Message-ID: <000c01cc3458$23838510$6a8a8f30$@com>

Hello friends!

 

Having trouble to start newly installed Kamailio

This is what I got:

 

root at debian:/home/user# /etc/init.d/kamailio status

Status of kamailio: kamailio is running.

root at debian:/home/user# kamctl start

 

INFO: Starting Kamailio : 

 

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start
failed

root at debian:/home/user#

 

Any ideas will appreciated

 

Regards

 

Roman

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From omar at 321communications.com  Mon Jun 27 06:43:54 2011
From: omar at 321communications.com (Omar)
Date: Mon, 27 Jun 2011 00:43:54 -0400
Subject: [SR-Users] MYSQL Compilation Kamailio 3.1.4
In-Reply-To: <4E0443A1.4070509@gmail.com>
References: <21FAF94F-ADCB-4714-8154-3D43E1308D0D@321communications.com>
	<4E0443A1.4070509@gmail.com>
Message-ID: <E41322A6-C2E2-4C00-8C1C-BDE97AC44C51@321communications.com>

added, and got same error:

bash-3.00# grep sin6 /usr/include/*
/usr/include/resolv.h:  struct sockaddr_in6     sin6;

Thanks Daniel

O

On Jun 24, 2011, at 3:58 AM, Daniel-Constantin Mierla wrote:

> Can you try adding
> 
> #include <netinet/in.h>
> 
> before first #include in modules/db_mysql/my_cmd.c file?
> 
> If not working, try to grep in /usr/include/ to see which is the file where sin6 is defined.
> 
> Cheers,
> Daniel
> 
> On 6/23/11 3:03 AM, Omar wrote:
>> bash-3.00# make clean
>> bash-3.00# make quiet=0
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c km_db_mysql.c -o km_db_mysql.o -MMD -MP
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_uri.c -o my_uri.o -MMD -MP
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_res.c -o my_res.o -MMD -MP
>> /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall    -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DDBG_QM_MALLOC -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 -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H  -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_cmd.c -o my_cmd.o -MMD -MP
>> In file included from ../../globals.h:37,
>>                  from ../../pt.h:42,
>>                  from ../../counters.h:43,
>>                  from mysql_mod.h:39,
>>                  from my_cmd.c:40:
>> ../../ip_addr.h:86: error: field 'sin6' has incomplete type
>> ../../ip_addr.h: In function 'ip_addr_loopback':
>> ../../ip_addr.h:283: warning: implicit declaration of function 'htonl'
>> ../../ip_addr.h:286: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK'
>> ../../ip_addr.h: In function 'sockaddr2ip_addr':
>> ../../ip_addr.h:353: error: dereferencing pointer to incomplete type
>> ../../ip_addr.h: In function 'su_getport':
>> ../../ip_addr.h:399: warning: implicit declaration of function 'ntohs'
>> ../../ip_addr.h: In function 'su_setport':
>> ../../ip_addr.h:418: warning: implicit declaration of function 'htons'
>> make: *** [my_cmd.o] Error 1
>> 
>> 
>> the onyl thing i did is uncomment the CROSSCOMPILE line
>> 
>> regards
>> 
>> 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
> 
> -- 
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
> 
> 
> 



From vaad.fabi at gmail.com  Mon Jun 27 10:03:08 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Mon, 27 Jun 2011 11:03:08 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E083770.1060906@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
	<4E008EFC.3020001@gmail.com> <4E044447.6030304@gmail.com>
	<4E045231.9080906@gmail.com> <4E04D395.4020709@gmail.com>
	<4E083770.1060906@gmail.com>
Message-ID: <4E08393C.9030205@gmail.com>

  Hi All and Daniel,


My interesting "realtime" story still continues :)

All configuration seems to be ok. It's default but some DB fields 
changes as Daniel advised.
Terminals registrations to kamailio is ok (but kamailio don't register 
as a endpoint on remote softswitch, but calls run :)  is it ok? )

Situation: one logic + two different terminals = two different results 
!!! Kamailio logic haven't changes (exclude DB fields) and it's not a 
softswitch part.
Is it something around two different requests with 
Proxy-Authenticate/realm (one case 193.58.XXX.XXX, second case 
VoipSwitch) and why kamailio generates two different realms ?



First case (ok).
Sip terminal - softphone twinkle
Dump from kamailio:
http://pastebin.com/uddxwf6F


Second case (not ok).
Sip terminal - cisco iphone
Dump from kamailio:

terminal->kamailio

     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 INVITE

     Max-Forwards: 70

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-1050babb;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.9e28

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.9e28

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 ACK

     Max-Forwards: 70

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 INVITE

     Max-Forwards: 70

     Proxy-Authorization: Digest        
username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-beb5344f;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     SIP/2.0 407 Proxy Authentication Required

     CSeq: 102 INVITE

     Via: SIP/2.0/UDP 
192.168.0.180:5060;rport=5060;received=93.116.XXX.XXX;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     To: <sip:9991 at 193.58.XXX.XXX>;tag=05050699909937

     Proxy-Authenticate: DIGEST realm="VoipSwitch", 
nonce="05f481310df9ef29-24050517506050625525"

     Content-Length: 0

     Record-Route: 
<sip:193.58.XXX.XXX;lr=on;ftag=dfe3d8151250ffabo0;nat=yes>


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=05050699909937

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 ACK

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 INVITE

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-303c9613;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.4a95

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.4a95

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 ACK

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0



-- 
Vadim F.

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From klaus.mailinglists at pernau.at  Mon Jun 27 10:10:19 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Mon, 27 Jun 2011 10:10:19 +0200
Subject: [SR-Users] changing script (perl, lua ...) without kamailio restart
Message-ID: <4E083AEB.9030807@pernau.at>

Hi!

Does any of the script interpreters (perl, lua, python ....) allow to
use a modified shell script without restarting Kamailio?

Thanks
Klaus



From miconda at gmail.com  Mon Jun 27 10:31:59 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Jun 2011 10:31:59 +0200
Subject: [SR-Users] new module: async - asynchronous SIP request processing
	in cfg
Message-ID: <4E083FFF.6020902@gmail.com>

Hello,

a new module is now in GIT master branch, named async.

Its purpose is to provide asynchronous SIP request processing using 
t_suspend()/t_continue() mechanism from tm module. While the tm module 
had it for quite some time, it was not available for usage in 
configuration file.

There are two functions implemented by now:

- async_route(routename, interval) - execute asynchronously a route 
block after a time interval
- aync_sleep(interval) - sleep asynchronously for a time interval and 
then resume SIP request processing

See the readme for more details:
http://kamailio.org/docs/modules/devel/modules/async.html

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Mon Jun 27 10:35:31 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Jun 2011 10:35:31 +0200
Subject: [SR-Users] changing script (perl,
	lua ...) without kamailio restart
In-Reply-To: <4E083AEB.9030807@pernau.at>
References: <4E083AEB.9030807@pernau.at>
Message-ID: <4E0840D3.90806@gmail.com>

Hello,

On 6/27/11 10:10 AM, Klaus Darilion wrote:
> Hi!
>
> Does any of the script interpreters (perl, lua, python ....) allow to
> use a modified shell script without restarting Kamailio?
lua_dofile() does that:
http://kamailio.org/docs/modules/stable/modules/app_lua.html#id2555384

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From vaad.fabi at gmail.com  Mon Jun 27 09:50:21 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Mon, 27 Jun 2011 10:50:21 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E04D395.4020709@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
	<4E008EFC.3020001@gmail.com> <4E044447.6030304@gmail.com>
	<4E045231.9080906@gmail.com> <4E04D395.4020709@gmail.com>
Message-ID: <4E08363D.9060909@gmail.com>

  Hi All and Daniel,


My interesting "realtime" story still continues :)

All configuration seems to be ok. It's default but some DB fields 
changes as Daniel advised.
Terminals registrations to kamailio is ok (but kamailio don't register 
as a endpoint on remote softswitch, but calls run :)  is it ok? )

Situation: one logic + two different terminals = two different results 
!!! Kamailio logic haven't changes (exclude DB fields) and it's not a 
softswitch part.
Is it something around two different requests with 
Proxy-Authenticate/realm (one case 193.58.XXX.XXX, second case 
VoipSwitch) and why kamailio generates two different realms ?



First case (ok).
Sip terminal - softphone twinkle
Dump from kamailio:

terminal->kamailio

     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKeweuqqbi

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 117 INVITE

     Contact: <sip:200 at 192.168.0.174:5065>

     Content-Type: application/sdp

     Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE

     Supported: replaces,norefersub,100rel

     User-Agent: Twinkle/1.4.2

     + SDP


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.174:5065;rport=5065;branch=z9hG4bKeweuqqbi;received=93.116.XXX.XXX

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.b6c5

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 117 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKeweuqqbi

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.b6c5

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 117 ACK

     User-Agent: Twinkle/1.4.2

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKtepuspep

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="200",realm="193.58.XXX.XXX",nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ",uri="sip:9991 at 193.58.XXX.XXX",response="e584c40d869c851f631c74c2a874cc02",algorithm=MD5

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 118 INVITE

     Contact: <sip:200 at 192.168.0.174:5065>

     Content-Type: application/sdp

     Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE

     Supported: replaces,norefersub,100rel

     User-Agent: Twinkle/1.4.2

     Content-Length: 311

     +SDP


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.174:5065;rport=5065;branch=z9hG4bKtepuspep;received=93.116.XXX.XXX

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 118 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     SIP/2.0 407 Proxy Authentication Required

     CSeq: 118 INVITE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKtepuspep

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050709100033375

     Proxy-Authenticate: DIGEST realm="VoipSwitch", 
nonce="05f6635f0e211fb3-24050517506070925525"

     Content-Length: 0

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKtepuspep

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050709100033375

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 118 ACK

     User-Agent: Twinkle/1.4.2

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKjgqxpmir

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="200",realm="193.58.XXX.XXX",nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ",uri="sip:9991 at 193.58.XXX.XXX",response="e584c40d869c851f631c74c2a874cc02",algorithm=MD5

     Proxy-Authorization: Digest 
username="200",realm="VoipSwitch",nonce="05f6635f0e211fb3-24050517506070925525",uri="sip:9991 at 193.58.XXX.XXX",response="c28d02e014adc119f75a839a7ebfb171",algorithm=MD5

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 119 INVITE

     Contact: <sip:200 at 192.168.0.174:5065>

     Content-Type: application/sdp

     Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE

     Supported: replaces,norefersub,100rel

     User-Agent: Twinkle/1.4.2

     Content-Length: 311

     +SDP


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.174:5065;rport=5065;branch=z9hG4bKjgqxpmir;received=93.116.XXX.XXX

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 119 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     IP/2.0 183 Session Progress

     CSeq: 119 INVITE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKjgqxpmir

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     Content-Type: application/sdp

     Content-Length: 226

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>

     +SDP

     SIP/2.0 200 OK

     CSeq: 119 INVITE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKjgqxpmir

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     Contact: <sip:93.116.XXX.XXX:5060;transport=udp>

     Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, 
MESSAGE, SUBSCRIBE, REFER

     Content-Type: application/sdp

     Content-Length: 226

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>


     RTP SESSION

     SIP/2.0 484 Address Incomplete

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-e8dcc9b4;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=b569480918f4e690o0

     To: <sip:193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.f963

     Call-ID: 2c9f671d-2c05328c at 192.168.0.180

     CSeq: 473 NOTIFY

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0

     RTP SESSION


     BYE sip:93.116.XXX.XXX:5060;transport=udp SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKepxrurdf

     Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 120 BYE

     User-Agent: Twinkle/1.4.2

     Content-Length: 0


     SIP/2.0 200 OK

     CSeq: 120 BYE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKepxrurdf

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     Content-Length: 0

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>



Second case (not ok).
Sip terminal - cisco iphone
Dump from kamailio:

terminal->kamailio

     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 INVITE

     Max-Forwards: 70

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-1050babb;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.9e28

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.9e28

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 ACK

     Max-Forwards: 70

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 INVITE

     Max-Forwards: 70

     Proxy-Authorization: Digest        
username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-beb5344f;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     SIP/2.0 407 Proxy Authentication Required

     CSeq: 102 INVITE

     Via: SIP/2.0/UDP 
192.168.0.180:5060;rport=5060;received=93.116.XXX.XXX;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     To: <sip:9991 at 193.58.XXX.XXX>;tag=05050699909937

     Proxy-Authenticate: DIGEST realm="VoipSwitch", 
nonce="05f481310df9ef29-24050517506050625525"

     Content-Length: 0

     Record-Route: 
<sip:193.58.XXX.XXX;lr=on;ftag=dfe3d8151250ffabo0;nat=yes>


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=05050699909937

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 ACK

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 INVITE

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-303c9613;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.4a95

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.4a95

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 ACK

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0
















On 06/24/2011 09:12 PM, vaad.fabi at gmail.com wrote:
>  Yes, NOTIFY was generated by ipphone.
>
> Thx, again :)
>
>
> On 06/24/2011 12:00 PM, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> based on the logs, the authentication fails due to wrong password. 
>> Check that you have the right password on client side.
>>
>> Kamailio does not generate any NOTIFY unless the presence server 
>> modules are loaded. If not, then the notify comes from somewhere else 
>> and it is just looped back due to config file routing logic.
>>
>> Cheers,
>> Daniel
>>
>> On 6/24/11 10:01 AM, vaad.fabi at gmail.com wrote:
>>>  Hi all,
>>>
>>> Still trying to run kamailio 3.1.x + asterisk 1.6.2.X realtime 
>>> architecture.
>>> As i mentioned earlier, instead of asterisk i use another softswitch 
>>> with db, table "retailroute" with "login" and "password" fields as 
>>> user auth data.
>>> Of course, troubles... user can't register to kamailio :(
>>>
>>> Logs( without users attempts to register, only kamailio internal logic:
>>>
>>>
>>> Jun 24 11:23:26 sbc kamailio[17768]: INFO: <script>: 
>>> ------------------ SIP Request: method [NOTIFY] from 
>>> [sip:100 at 100.100.100.100] to [sip:100.100.100.100]
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <811> <NOTIFY>
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: maxfwd [mf_funcs.c:85]: 
>>> value = 70
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [parser/msg_parser.c:103]: found end of header
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [msg_translator.c:204]: check_via_address(XX.116.255.XXX, 
>>> 192.168.0.180, 0)
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: siputils [checks.c:73]: 
>>> no totag
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1081]: 
>>> DEBUG: t_check_msg: msg id=39 global id=38 T start=(nil)
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:528]: 
>>> t_lookup_request: start searching: hash=31430, isACK=0
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:485]: 
>>> DEBUG: RFC3261 transaction matching failed
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:711]: 
>>> DEBUG: t_lookup_request: no transaction found
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: tm [t_lookup.c:1150]: 
>>> DEBUG: t_check_msg: msg id=39 global id=39 T end=(nil)
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [select.c:424]: 
>>> Calling SELECT 0x832b7e4
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==13 
>>> &&  [100.100.100.100] == [100.100.100.100]
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches 
>>> port 5060
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db 
>>> [authorize.c:239]: realm value [100.100.100.100]
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:95]: auth: 
>>> digest-algo: MD5 parsed value: 1
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:118]: 
>>> allocate 28 bytes for result set at 0x832c8cc
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:68]: 
>>> 1 columns returned from the query
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:155]: 
>>> allocate 4 bytes for result names at 0x832ba40
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:165]: 
>>> allocate 4 bytes for result types at 0x832ba08
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:85]: 
>>> allocate 8 bytes for RES_NAMES[0] at 0x832c950
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:92]: 
>>> RES_NAMES(0x832c950)[0]=[password]
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: db_mysql [km_res.c:136]: 
>>> use DB1_STRING result type
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:184]: 
>>> allocate 8 bytes for rows at 0x832c918
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:119]: 
>>> allocate 20 bytes for row values at 0x832e75c
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_val.c:117]: 
>>> converting STRING [dav0011]
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth_db 
>>> [authorize.c:120]: HA1 string calculated: 
>>> a8fdf502e497ff8e9407ac2dc31540df
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:205]: 
>>> check_response: Our result = 'fa83bf7c0a7af33cc500aa2b62758931'
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [api.c:215]: 
>>> check_response: Authorization failed
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:81]: 
>>> freeing 1 columns
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:85]: 
>>> freeing RES_NAMES[0] at 0x832c950
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:94]: 
>>> freeing result names at 0x832ba40
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:99]: 
>>> freeing result types at 0x832ba08
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:54]: 
>>> freeing 1 rows
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_row.c:97]: 
>>> freeing row values at 0x832e75c
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:62]: 
>>> freeing rows at 0x832c918
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [db_res.c:136]: 
>>> freeing result set at 0x832c8cc
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:102]: 
>>> build_challenge_hf: realm='100.100.100.100'
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: auth [challenge.c:236]: 
>>> auth: 'Proxy-Authenticate: Digest realm="100.100.100.100", 
>>> nonce="TgQ8mU4EO22Odd1W63tOO27IRrKX0kU '
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: sl [sl.c:278]: reply in 
>>> stateless mode (sl)
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> 
>>> [msg_translator.c:204]: check_via_address(XX.116.255.XXX, 
>>> 192.168.0.180, 0)
>>> Jun 24 11:23:26 sbc kamailio[17768]: DEBUG: <core> [usr_avp.c:646]: 
>>> DEBUG:destroy_avp_list: destroying list (nil)
>>>
>>>
>>>
>>> Such messages periodically sends kamailio. Why kamailio every time 
>>> sends yourself these NOTIFY messages, what it try to do ?
>>> I see problem something around auth_db autorization and HA1, isn't 
>>> it ? How to fix it ?
>>>
>>>
>>
>
>


-- 
Vadim F.

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From vaad.fabi at gmail.com  Mon Jun 27 09:55:28 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Mon, 27 Jun 2011 10:55:28 +0300
Subject: [SR-Users] kamailio realtime integration
In-Reply-To: <4E04D395.4020709@gmail.com>
References: <4DFF4DFF.1070602@gmail.com>	<AB4F45F892DD44FF8E46CA78F03BCE8E@work>
	<4E0055E7.6030808@gmail.com> <4E005E62.3090205@gmail.com>
	<4E006A8C.50704@gmail.com> <4E006FCA.9010904@gmail.com>
	<4E008EFC.3020001@gmail.com> <4E044447.6030304@gmail.com>
	<4E045231.9080906@gmail.com> <4E04D395.4020709@gmail.com>
Message-ID: <4E083770.1060906@gmail.com>

  Hi All and Daniel,


My interesting "realtime" story still continues :)

All configuration seems to be ok. It's default but some DB fields 
changes as Daniel advised.
Terminals registrations to kamailio is ok (but kamailio don't register 
as a endpoint on remote softswitch, but calls run :)  is it ok? )

Situation: one logic + two different terminals = two different results 
!!! Kamailio logic haven't changes (exclude DB fields) and it's not a 
softswitch part.
Is it something around two different requests with 
Proxy-Authenticate/realm (one case 193.58.XXX.XXX, second case 
VoipSwitch) and why kamailio generates two different realms ?



First case (ok).
Sip terminal - softphone twinkle
Dump from kamailio:

terminal->kamailio

     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKeweuqqbi

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 117 INVITE

     Contact: <sip:200 at 192.168.0.174:5065>

     Content-Type: application/sdp

     Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE

     Supported: replaces,norefersub,100rel

     User-Agent: Twinkle/1.4.2

     + SDP


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.174:5065;rport=5065;branch=z9hG4bKeweuqqbi;received=93.116.XXX.XXX

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.b6c5

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 117 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKeweuqqbi

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.b6c5

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 117 ACK

     User-Agent: Twinkle/1.4.2

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKtepuspep

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="200",realm="193.58.XXX.XXX",nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ",uri="sip:9991 at 193.58.XXX.XXX",response="e584c40d869c851f631c74c2a874cc02",algorithm=MD5

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 118 INVITE

     Contact: <sip:200 at 192.168.0.174:5065>

     Content-Type: application/sdp

     Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE

     Supported: replaces,norefersub,100rel

     User-Agent: Twinkle/1.4.2

     Content-Length: 311

     +SDP


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.174:5065;rport=5065;branch=z9hG4bKtepuspep;received=93.116.XXX.XXX

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 118 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     SIP/2.0 407 Proxy Authentication Required

     CSeq: 118 INVITE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKtepuspep

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050709100033375

     Proxy-Authenticate: DIGEST realm="VoipSwitch", 
nonce="05f6635f0e211fb3-24050517506070925525"

     Content-Length: 0

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKtepuspep

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050709100033375

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 118 ACK

     User-Agent: Twinkle/1.4.2

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKjgqxpmir

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="200",realm="193.58.XXX.XXX",nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ",uri="sip:9991 at 193.58.XXX.XXX",response="e584c40d869c851f631c74c2a874cc02",algorithm=MD5

     Proxy-Authorization: Digest 
username="200",realm="VoipSwitch",nonce="05f6635f0e211fb3-24050517506070925525",uri="sip:9991 at 193.58.XXX.XXX",response="c28d02e014adc119f75a839a7ebfb171",algorithm=MD5

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 119 INVITE

     Contact: <sip:200 at 192.168.0.174:5065>

     Content-Type: application/sdp

     Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE

     Supported: replaces,norefersub,100rel

     User-Agent: Twinkle/1.4.2

     Content-Length: 311

     +SDP


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.174:5065;rport=5065;branch=z9hG4bKjgqxpmir;received=93.116.XXX.XXX

     To: <sip:9991 at 193.58.XXX.XXX>

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 119 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     IP/2.0 183 Session Progress

     CSeq: 119 INVITE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKjgqxpmir

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     Content-Type: application/sdp

     Content-Length: 226

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>

     +SDP

     SIP/2.0 200 OK

     CSeq: 119 INVITE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKjgqxpmir

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     Contact: <sip:93.116.XXX.XXX:5060;transport=udp>

     Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, 
MESSAGE, SUBSCRIBE, REFER

     Content-Type: application/sdp

     Content-Length: 226

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>


     RTP SESSION

     SIP/2.0 484 Address Incomplete

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-e8dcc9b4;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=b569480918f4e690o0

     To: <sip:193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.f963

     Call-ID: 2c9f671d-2c05328c at 192.168.0.180

     CSeq: 473 NOTIFY

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0

     RTP SESSION


     BYE sip:93.116.XXX.XXX:5060;transport=udp SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKepxrurdf

     Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>

     Max-Forwards: 70

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     CSeq: 120 BYE

     User-Agent: Twinkle/1.4.2

     Content-Length: 0


     SIP/2.0 200 OK

     CSeq: 120 BYE

     Via: SIP/2.0/UDP 
192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKepxrurdf

     From: "kamailio" <sip:200 at 193.58.XXX.XXX>;tag=wvgqh

     Call-ID: lutmonzrdvxboxk at SipuraSPA

     To: <sip:9991 at 193.58.XXX.XXX>;tag=050710100033890

     Content-Length: 0

     Record-Route: <sip:193.58.XXX.XXX;lr=on;ftag=wvgqh;nat=yes>



Second case (not ok).
Sip terminal - cisco iphone
Dump from kamailio:

terminal->kamailio

     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 INVITE

     Max-Forwards: 70

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-1050babb;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.9e28

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.9e28

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 101 ACK

     Max-Forwards: 70

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 INVITE

     Max-Forwards: 70

     Proxy-Authorization: Digest        
username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 100 trying -- your call is important to us

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-beb5344f;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 INVITE

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


     SIP/2.0 407 Proxy Authentication Required

     CSeq: 102 INVITE

     Via: SIP/2.0/UDP 
192.168.0.180:5060;rport=5060;received=93.116.XXX.XXX;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     To: <sip:9991 at 193.58.XXX.XXX>;tag=05050699909937

     Proxy-Authenticate: DIGEST realm="VoipSwitch", 
nonce="05f481310df9ef29-24050517506050625525"

     Content-Length: 0

     Record-Route: 
<sip:193.58.XXX.XXX;lr=on;ftag=dfe3d8151250ffabo0;nat=yes>


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=05050699909937

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 102 ACK

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0


     INVITE sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 INVITE

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     Expires: 240

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 259

     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

     Supported: replaces

     Content-Type: application/sdp


kamailio->terminal

     SIP/2.0 407 Proxy Authentication Required

     Via: SIP/2.0/UDP 
192.168.0.180:5060;branch=z9hG4bK-303c9613;rport=5060;received=93.116.XXX.XXX

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.4a95

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 INVITE

     Proxy-Authenticate: Digest realm="193.58.XXX.XXX", 
nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE"

     Server: kamailio (3.1.4 (i386/linux))

     Content-Length: 0


terminal->kamailio

     ACK sip:9991 at 193.58.XXX.XXX SIP/2.0

     Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613

     From: "100" <sip:100 at 193.58.XXX.XXX>;tag=dfe3d8151250ffabo0

     To: <sip:9991 at 193.58.XXX.XXX>;tag=636f77b6676bf8d50aad9baa13cd3627.4a95

     Call-ID: 8760c7dd-41ef9283 at 192.168.0.180

     CSeq: 103 ACK

     Max-Forwards: 70

     Proxy-Authorization: Digest 
username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9991 at 193.58.XXX.XXX",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732"

     Contact: "100" <sip:100 at 192.168.0.180:5060>

     User-Agent: Linksys/SPA922-5.1.15(a)

     Content-Length: 0

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From miconda at gmail.com  Mon Jun 27 10:48:34 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Jun 2011 10:48:34 +0200
Subject: [SR-Users] async processing in a module
In-Reply-To: <BANLkTi=NqNeeGp+E1jRQrY_M-fNK4fGBdQ@mail.gmail.com>
References: <BANLkTi=NqNeeGp+E1jRQrY_M-fNK4fGBdQ@mail.gmail.com>
Message-ID: <4E0843E2.8090102@gmail.com>

Hello,

just to close this thread. A new module named async has been introduced 
in GIT master, showing how to use the tm functions for asynchronous sip 
request processing.

It should answer the questions posted in this message.

Cheers,
Daniel

On 6/24/11 1:00 PM, Matthew Williams wrote:
> Hi,
>
> I'm working on a jsonrpc (client) module that we would like to make 
> async, using t_suspend and t_continue from the tm module. I have a 
> couple questions related to this.
>
> The way I understand these functions, t_suspend will freeze the 
> transaction in shared memory and immediately return and t_continue 
> will unfreeze the transaction. What I'm not clear on here is whether 
> t_continue will continue processing of the transaction within the 
> process that calls t_continue, or will it delegate to a worker process?
>
> I assume I should spawn an io process in child_init which will be 
> responsible for sending/receiving on the socket(s), but what is the 
> best way to communicate between processes? Is there a way to get a fd 
> from fork_process (that can be written to by any child and read from 
> using select() in the io thread)?
>
> This is my first attempt at a real module, so any pointers are greatly 
> appreciated.
>
> Regards,
> Matthew WIlliams
>
>
> _______________________________________________
> 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://linkedin.com/in/miconda -- http://twitter.com/miconda

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From ibc at aliax.net  Mon Jun 27 11:44:00 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 27 Jun 2011 11:44:00 +0200
Subject: [SR-Users] async processing in a module
In-Reply-To: <4E0843E2.8090102@gmail.com>
References: <BANLkTi=NqNeeGp+E1jRQrY_M-fNK4fGBdQ@mail.gmail.com>
	<4E0843E2.8090102@gmail.com>
Message-ID: <BANLkTi=M5f-j2NVmiz6NupnvOfn8cyc3pQ@mail.gmail.com>

2011/6/27 Daniel-Constantin Mierla <miconda at gmail.com>:
> just to close this thread. A new module named async has been introduced in
> GIT master, showing how to use the tm functions for asynchronous sip request
> processing.

Hi Daniel, good addition. Let me a question, the module description says:

--------------------
Note that after invoking the an asyncronous operation, the processing
will continue later, in another application process. Therefore, do not
rely on variables stored in private memory, used shared memory if you
want to get values after the processing is resumend (e.g., $shv(...)
of htable $sht(...)).
---------------------

I assume that using AVP's is also valid (as AVP's are stored within
the transaction), perhaps the doc could also mention them?

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


From abalashov at evaristesys.com  Mon Jun 27 11:47:38 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 27 Jun 2011 05:47:38 -0400
Subject: [SR-Users] new module: async - asynchronous SIP request
 processing in cfg
In-Reply-To: <4E083FFF.6020902@gmail.com>
References: <4E083FFF.6020902@gmail.com>
Message-ID: <4E0851BA.6040202@evaristesys.com>

Interesting.  A few thoughts:

1. What sort of use cases are envisioned here?  Most SIP transactions 
are very delay-sensitive operate on a sub-1s retransmit timer 
resolution, e.g. Timer T1.

For INVITE transactions where retransmissions can be held down by 100 
Trying, I am still brought to wonder, what is the end goal, given that 
in practice, pretty much all blocking calls invoked in processing an 
INVITE request return info that is essential to further processing/routing?

I have a feeling that perhaps this is more useful with SIP features that 
are not "call-related", but I am not sufficiently imaginative to come up 
with a concrete example.

Oddly enough, one constituency that I do think would have an interest in 
this feature--though a dubiously desirable one--is people running large 
amounts of low-ASR, low-ACD dialer traffic.  We get asked a lot about 
whether there are ways to "slow down" outbound BYEs (from the equipment 
as they move toward a termination provider) in order to circumvent 
financial penalties assessed when a percentage threshold of 
short-duration calls (e.g. under 6 seconds) is exceeded.

Despite the obvious technical problems that can potentially be incurred 
here (retransmissions, fast RTP timeout, etc.), doubtless this is going 
to excite garbage traffic runners.

I am not sure that's a good thing, though.  The PSTN is like the sewer 
system;  you get out of it what you put into it.  And what these people 
put into it in fact belongs in a sewer system.  :-)

2. It would be nice if the interval granularity could be expressed in 
milliseconds, since that would allow potential applications that fly 
under the radar of most SIP timers.

3. It seems like in some ways this would be most useful if blocking 
operations like database calls could be "redirected" into a separate 
worker thread pool, and then request processing would resume as soon as 
they complete.  Because all that happens in separate threads, it frees 
up core SIP worker threads to absorb more new requests.

So, perhaps something like:

     route {
        ...
        async_wrapper(allow_trusted(), MY_ROUTE);
     }

     route[MY_ROUTE] {
        if(!$someretval) {
          sl_send_reply("403", "Forbidden");
          exit;
        }

        t_relay(); # etc.
     }

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From zhulizhong at live.com  Mon Jun 27 13:14:24 2011
From: zhulizhong at live.com (James zhu)
Date: Mon, 27 Jun 2011 11:14:24 +0000
Subject: [SR-Users] Kamailio start failed
In-Reply-To: <000c01cc3458$23838510$6a8a8f30$@com>
References: <000c01cc3458$23838510$6a8a8f30$@com>
Message-ID: <blu121-w14D479D8941E853ECB5396B5570@phx.gbl>


try reboot the server.

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 


From: roman at dmytriv.com
To: sr-users at lists.sip-router.org
Date: Sun, 26 Jun 2011 19:24:03 -0400
Subject: [SR-Users] Kamailio start failed



Hello friends! Having trouble to start newly installed KamailioThis is what I got: root at debian:/home/user# /etc/init.d/kamailio statusStatus of kamailio: kamailio is running.root at debian:/home/user# kamctl start INFO: Starting Kamailio :  ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failedroot at debian:/home/user# Any ideas will appreciated Regards Roman
_______________________________________________
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 vaad.fabi at gmail.com  Mon Jun 27 15:02:45 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Mon, 27 Jun 2011 16:02:45 +0300
Subject: [SR-Users] realtime  asterisk double-authentication
In-Reply-To: <blu121-w14D479D8941E853ECB5396B5570@phx.gbl>
References: <000c01cc3458$23838510$6a8a8f30$@com>
	<blu121-w14D479D8941E853ECB5396B5570@phx.gbl>
Message-ID: <4E087F75.1010306@gmail.com>

 Hi all,

> By default, Asterisk uses the column *secret* for SIP user password,
but if that is filled in,
> Asterisk will ask for authentication again, resulting in
double-authentication which we want to avoid.

What does "double-authentication" mean ? Is it standart  "INVITE, 407+nonce, ACK, INVITE+nonce+responce" scheme ?

-- 
Vadim F.

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From roman at dmytriv.com  Mon Jun 27 15:31:06 2011
From: roman at dmytriv.com (Roman Dmytriv)
Date: Mon, 27 Jun 2011 09:31:06 -0400
Subject: [SR-Users] Kamailio start failed
In-Reply-To: <blu121-w14D479D8941E853ECB5396B5570@phx.gbl>
References: <000c01cc3458$23838510$6a8a8f30$@com>
	<blu121-w14D479D8941E853ECB5396B5570@phx.gbl>
Message-ID: <000f01cc34ce$78a61780$69f24680$@com>

Thank you but that actually didn't help. 

What I found is that kamailio.pid did exist in /var/run/Kamailio/. So, I
copied it to /var/run and kamctl started automatically.

 

Now I can register client with the server but only using IP not domain. 

Domain resolves to IP correctly.

/etc/kamailio/kamctlrc is set

SIP_DOMAIN=voip.mydomain.com

What can prevent client from using domain?

 

Regards

 

Roman

 

From: James zhu [mailto:zhulizhong at live.com] 
Sent: Monday, June 27, 2011 7:14 AM
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Kamailio start failed

 

try reboot the server.

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 



  _____  

From: roman at dmytriv.com
To: sr-users at lists.sip-router.org
Date: Sun, 26 Jun 2011 19:24:03 -0400
Subject: [SR-Users] Kamailio start failed

Hello friends!

 

Having trouble to start newly installed Kamailio

This is what I got:

 

root at debian:/home/user# /etc/init.d/kamailio status

Status of kamailio: kamailio is running.

root at debian:/home/user# kamctl start

 

INFO: Starting Kamailio : 

 

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start
failed

root at debian:/home/user#

 

Any ideas will appreciated

 

Regards

 

Roman


_______________________________________________ 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 fborot at hotmail.com  Mon Jun 27 16:40:30 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Mon, 27 Jun 2011 10:40:30 -0400
Subject: [SR-Users] kamailio fails to start with kamctl start
In-Reply-To: <SNT110-W4593A88C65D2250B77EE35CD520@phx.gbl>
References: <SNT110-W64C60382EE6B2F2796781CD520@phx.gbl>,
	<SNT110-W4593A88C65D2250B77EE35CD520@phx.gbl>
Message-ID: <SNT110-W9414B437AC3ED92BC2736CD570@phx.gbl>



I fixed the problem but I would like somebody to shed some light into this issue.

I noticed that when I run kamailio w/o telling where to load the kamailio.cfg file is when it failed. So I went to the kamctl script and notice that in the section where "kamctl start" happens the variable $STARTOPTIONS was empty. I went to the etc/kamailio/kamctlrc, un-commented the line with $STARTOPTIONS and made it like this:


 
STARTOPTIONS=" -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg" -> this points to the config file which is in a custom path

So, it seems that when one installs kamailio in a directory other than the default path with the prefix option ['make prefix="new path here" all' and 'make prefix="new path here" install'] you have to explicitly tell it somehow to load the config file from somewhere else. Or I did not do it correctly. Can somebody please confirm?.
I thought that the initialization scripts would also take care of that. I have installed kamailio 1.4.3  1.5.0, 1.5.2 using the prefix option at 'make' time several times and never had this issue.

Again, these were my main steps:

 mkdir -p /usr/local/src/kamailio-3.1.0
cd /usr/local/src/kamailio-3.1.0

git clone --depth 1 git://git.sip-router.org/sip-router kamailio
cd kamailio
git checkout -b 3.1 origin/3.1

mkdir /usr/local/kamailio-3.1

make FLAVOUR=kamailio cfg

vi modules.lst to exclude db_mysql db_odbc from modules.lst 

make PREFIX="/usr/local/kamailio-3.1/" all
make PREFIX="/usr/local/kamailio-3.1/" install


then I executed /usr/local/kamailio-3.1/sbin/kamdbctl create to create the DB and tables.



From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Date: Fri, 24 Jun 2011 18:10:38 -0400
Subject: Re: [SR-Users] kamailio fails to start with kamctl start









I enabled debugging on the kamctl script and see these lines right before it fails:

+ minfo 'Starting Kamailio : '
+ '[' -t 1 -a -z '' ']'
+ echo -e '\E[37;33m\033[1mINFO: Starting Kamailio : \033[0m'
INFO: Starting Kamailio :
+ '[' -r /var/run/kamailio.pid ']'
+ '[' '!' -x ./kamailio ']'
+ '[' 1 = 1 ']'
+ ./kamailio -P /var/run/kamailio.pid
+ sleep 3
+ '[' '!' -s /var/run/kamailio.pid ']'
+ echo

+ merr 'PID file /var/run/kamailio.pid does not exist -- Kamailio start failed'
+ '[' -t 1 -a -z '' ']'
+ echo -e '\E[37;31m\033[1mERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed\033[0m'
ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
+ exit 1
[root at registrar sbin]#


these are the main steps I followed to install it:

mkdir -p /usr/local/src/kamailio-3.1.0
cd /usr/local/src/kamailio-3.1.0

git clone --depth 1 git://git.sip-router.org/sip-router kamailio
cd kamailio
git checkout -b 3.1 origin/3.1

mkdir /usr/local/kamailio-3.1

make FLAVOUR=kamailio cfg

vi modules.lst to exclude db_mysql db_odbc from modules.lst 

make PREFIX="/usr/local/kamailio-3.1/" all
make PREFIX="/usr/local/kamailio-3.1/" install


then I executed /usr/local/kamailio-3.1/sbin/kamdbctl create to create the DB and tables.



From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Date: Fri, 24 Jun 2011 16:28:28 -0400
Subject: [SR-Users] kamailio fails to start with kamctl start








Hello
I just installed kamailio-3.1 and using the default config file "kamctl start" fails


 [root at registrar sbin]# ./kamctl start

INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
[root at registrar sbin]# 

but when I use this command it starts: <kamailio -f path_to_config_file

[root at registrar sbin]# ./kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg
loading modules under /usr/local/kamailio-3.1//lib64/kamailio/modules_k/:/usr/local/kamailio-3.1//lib64/kamailio/modules/
Listening on
             udp: 127.0.0.1:5060
             udp: 192.168.168.10:5060
             tcp: 127.0.0.1:5060
             tcp: 192.168.168.10:5060
Aliases:
             tcp: localhost:5060
             tcp: localhost.localdomain:5060
             tcp: registrar:5060
             tcp: registrar.mytestdomain.com:5060
             udp: localhost:5060
             udp: localhost.localdomain:5060
             udp: registrar:5060
             udp: registrar.mytestdomain.com:5060

[root at registrar sbin]#

Also, when I use "./kamctl start" the syslog file does not show anything at all that would help me why it is failing. 
But when I use "/usr/local/kamailio-3.1/sbin/kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg" it does update the syslog file.

Also, after kamailio start with "./kamailio -f .." I can use ./kamctl fifo <any option>. Example:

[root at registrar sbin]# ./kamctl fifo version
Server:: kamailio (3.1.4 (x86_64/linux))
Build:: mi_core.c compiled on 15:09:40 Jun 24 2011 with gcc 4.1.2
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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
GIT:: e2d6f0
[root at registrar sbin]#

[root at registrar sbin]# ./kamctl fifo ps
Process::  ID=0 PID=27078 Type=attendant
Process::  ID=1 PID=27079 Type=udp receiver child=0 sock=127.0.0.1:5060
Process::  ID=2 PID=27080 Type=udp receiver child=1 sock=127.0.0.1:5060
Process::  ID=3 PID=27081 Type=udp receiver child=2 sock=127.0.0.1:5060
Process::  ID=4 PID=27082 Type=udp receiver child=3 sock=127.0.0.1:5060
Process::  ID=5 PID=27083 Type=udp receiver child=0 sock=192.168.168.10:5060
Process::  ID=6 PID=27084 Type=udp receiver child=1 sock=192.168.168.10:5060
Process::  ID=7 PID=27085 Type=udp receiver child=2 sock=192.168.168.10:5060
Process::  ID=8 PID=27086 Type=udp receiver child=3 sock=192.168.168.10:5060
Process::  ID=9 PID=27087 Type=slow timer
Process::  ID=10 PID=27088 Type=timer
Process::  ID=11 PID=27089 Type=MI FIFO
Process::  ID=12 PID=27090 Type=ctl handler
Process::  ID=13 PID=27091 Type=tcp receiver child=0
Process::  ID=14 PID=27092 Type=tcp receiver child=1
Process::  ID=15 PID=27093 Type=tcp receiver child=2
Process::  ID=16 PID=27094 Type=tcp receiver child=3
Process::  ID=17 PID=27095 Type=tcp main process
[root at registrar sbin]#



 		 	   		  

_______________________________________________
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 miconda at gmail.com  Mon Jun 27 16:50:48 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Jun 2011 16:50:48 +0200
Subject: [SR-Users] async processing in a module
In-Reply-To: <BANLkTi=M5f-j2NVmiz6NupnvOfn8cyc3pQ@mail.gmail.com>
References: <BANLkTi=NqNeeGp+E1jRQrY_M-fNK4fGBdQ@mail.gmail.com>
	<4E0843E2.8090102@gmail.com>
	<BANLkTi=M5f-j2NVmiz6NupnvOfn8cyc3pQ@mail.gmail.com>
Message-ID: <4E0898C8.40800@gmail.com>



On 6/27/11 11:44 AM, I?aki Baz Castillo wrote:
> 2011/6/27 Daniel-Constantin Mierla<miconda at gmail.com>:
>> just to close this thread. A new module named async has been introduced in
>> GIT master, showing how to use the tm functions for asynchronous sip request
>> processing.
> Hi Daniel, good addition. Let me a question, the module description says:
>
> --------------------
> Note that after invoking the an asyncronous operation, the processing
> will continue later, in another application process. Therefore, do not
> rely on variables stored in private memory, used shared memory if you
> want to get values after the processing is resumend (e.g., $shv(...)
> of htable $sht(...)).
> ---------------------
>
> I assume that using AVP's is also valid (as AVP's are stored within
> the transaction), perhaps the doc could also mention them?
Hi Inaki,

yes, transaction avps and xavps should be also available when resuming 
the processing. You can add a phrase to explain that at your convenience.

Thanks,
Daniel
-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From miconda at gmail.com  Mon Jun 27 17:10:34 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 27 Jun 2011 17:10:34 +0200
Subject: [SR-Users] new module: async - asynchronous SIP request
 processing in cfg
In-Reply-To: <4E0851BA.6040202@evaristesys.com>
References: <4E083FFF.6020902@gmail.com> <4E0851BA.6040202@evaristesys.com>
Message-ID: <4E089D6A.7040605@gmail.com>

Hi Alex,

On 6/27/11 11:47 AM, Alex Balashov wrote:
> Interesting.  A few thoughts:
>
> 1. What sort of use cases are envisioned here?  Most SIP transactions 
> are very delay-sensitive operate on a sub-1s retransmit timer 
> resolution, e.g. Timer T1.

the first purpose what to show how to write C extensions using the 
asynchronous SIP request processing mechanism, based on mailing list 
questions lately.

In the past there were some discussions about asynchronous sleep and 
this was used as a sample implementation, by executiing either next 
actions in config (with some limitations listed in readme) or a specific 
route block when processing is resumed.


>
> For INVITE transactions where retransmissions can be held down by 100 
> Trying, I am still brought to wonder, what is the end goal, given that 
> in practice, pretty much all blocking calls invoked in processing an 
> INVITE request return info that is essential to further 
> processing/routing?
>
> I have a feeling that perhaps this is more useful with SIP features 
> that are not "call-related", but I am not sufficiently imaginative to 
> come up with a concrete example.

There are also use cases for INVITEs as well, by conding some extension 
in C. Matthew Williams showed a case, when it is needed to connect to 
another server before allowing the call.

>
> Oddly enough, one constituency that I do think would have an interest 
> in this feature--though a dubiously desirable one--is people running 
> large amounts of low-ASR, low-ACD dialer traffic.  We get asked a lot 
> about whether there are ways to "slow down" outbound BYEs (from the 
> equipment as they move toward a termination provider) in order to 
> circumvent financial penalties assessed when a percentage threshold of 
> short-duration calls (e.g. under 6 seconds) is exceeded.
>
> Despite the obvious technical problems that can potentially be 
> incurred here (retransmissions, fast RTP timeout, etc.), doubtless 
> this is going to excite garbage traffic runners.
Sending 100 reply should stop retransmission for any kind of SIP request.

>
> I am not sure that's a good thing, though.  The PSTN is like the sewer 
> system;  you get out of it what you put into it.  And what these 
> people put into it in fact belongs in a sewer system.  :-)
>
> 2. It would be nice if the interval granularity could be expressed in 
> milliseconds, since that would allow potential applications that fly 
> under the radar of most SIP timers.

The easiest so far was to use a generic timer api which is second 
granularity, but further improvements are planned.

>
> 3. It seems like in some ways this would be most useful if blocking 
> operations like database calls could be "redirected" into a separate 
> worker thread pool, and then request processing would resume as soon 
> as they complete.  Because all that happens in separate threads, it 
> frees up core SIP worker threads to absorb more new requests.

This is kind of next improvement, something like using an internal 
message queue to pass transaction identifiers (returned by t_suspend()) 
to other processes that will resume the processing immediately by 
executing further a specific route block. I am not looking to create a 
generic async wrapper right now to cfg functions, that will require some 
analysis to see the feasibility (i.e., passing a function name with 
parameters as parameter to another cfg function might give some troubles 
with our internal fixup system). But directing the processing of a route 
block to another pool of workers is more or less the same (if you have 
just that function executed in the route block).

Cheers,
Daniel
>
> So, perhaps something like:
>
>     route {
>        ...
>        async_wrapper(allow_trusted(), MY_ROUTE);
>     }
>
>     route[MY_ROUTE] {
>        if(!$someretval) {
>          sl_send_reply("403", "Forbidden");
>          exit;
>        }
>
>        t_relay(); # etc.
>     }
>

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda



From ibc at aliax.net  Mon Jun 27 17:10:45 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Mon, 27 Jun 2011 17:10:45 +0200
Subject: [SR-Users] async processing in a module
In-Reply-To: <4E0898C8.40800@gmail.com>
References: <BANLkTi=NqNeeGp+E1jRQrY_M-fNK4fGBdQ@mail.gmail.com>
	<4E0843E2.8090102@gmail.com>
	<BANLkTi=M5f-j2NVmiz6NupnvOfn8cyc3pQ@mail.gmail.com>
	<4E0898C8.40800@gmail.com>
Message-ID: <BANLkTin1OskXxaVjQqbpEwGtket-5f0aug@mail.gmail.com>

2011/6/27 Daniel-Constantin Mierla <miconda at gmail.com>:
> Hi Inaki,
>
> yes, transaction avps and xavps should be also available when resuming the
> processing. You can add a phrase to explain that at your convenience.

Ok, added.

Thanks a lot.

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


From fborot at hotmail.com  Mon Jun 27 17:45:32 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Mon, 27 Jun 2011 11:45:32 -0400
Subject: [SR-Users] LCR module
Message-ID: <SNT110-W598AA8AF2312A4A301DB7BCD570@phx.gbl>





After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time:



[root at registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/
[root at registrar modules_k]# ls
acc.so            db_text.so       drouting.so  mi_datagram.so    permissions.so  rtimer.so     statistics.so     usrloc.so
alias_db.so       db_unixodbc.so   exec.so      mi_fifo.so        pike.so         rtpproxy.so   textops.so        xlog.so
auth_db.so        dialog.so        group.so     msilo.so          pua_mi.so       siptrace.so   tmx.so
auth_diameter.so  dispatcher.so    htable.so    nathelper.so      pv.so           siputils.so   uac_redirect.so
benchmark.so      diversion.so     imc.so       nat_traversal.so  qos.so          speeddial.so  uac.so
call_control.so   domainpolicy.so  kex.so       path.so           registrar.so    sqlops.so     uri_db.so
cfgutils.so       domain.so        maxfwd.so    pdt.so            rr.so           sst.so        userblacklist.so

[root at registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/
[root at registrar modules_s]# ls
acc_db.so      avp.so         diversion.so  fifo.so    nathelper.so    pike.so       speeddial.so  uri_db.so
acc_syslog.so  db_ops.so      domain.so     gflags.so  options.so      print.so      textops.so    uri.so
auth_db.so     dialog.so      eval.so       maxfwd.so  pdt.so          registrar.so  timer.so      usrloc.so
avp_db.so      dispatcher.so  exec.so       msilo.so   permissions.so  rr.so         uac.so        xlog.so
[root at registrar modules_s]#

I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something?

This is from the syslog

Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume

and this is from the fifo interface:

[root at registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload
500 command 'lcr' not available
[root at registrar kamailio]#

 		 	   		  
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From oej at edvina.net  Mon Jun 27 16:11:25 2011
From: oej at edvina.net (Olle E. Johansson)
Date: Mon, 27 Jun 2011 16:11:25 +0200
Subject: [SR-Users] [sr-dev] new module: async - asynchronous SIP
	request processing in cfg
In-Reply-To: <4E083FFF.6020902@gmail.com>
References: <4E083FFF.6020902@gmail.com>
Message-ID: <870716D7-124A-4F66-B770-D523BDE6B7E6@edvina.net>


27 jun 2011 kl. 10.31 skrev Daniel-Constantin Mierla:

> Hello,
> 
> a new module is now in GIT master branch, named async.
> 
> Its purpose is to provide asynchronous SIP request processing using t_suspend()/t_continue() mechanism from tm module. While the tm module had it for quite some time, it was not available for usage in configuration file.
> 
> There are two functions implemented by now:
> 
> - async_route(routename, interval) - execute asynchronously a route block after a time interval
> - aync_sleep(interval) - sleep asynchronously for a time interval and then resume SIP request processing
> 
> See the readme for more details:
> http://kamailio.org/docs/modules/devel/modules/async.html
> 
Seems kind of sexy.

Do you have a good example of usage?

Greetings from Madrid
/O



From fborot at hotmail.com  Mon Jun 27 17:57:05 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Mon, 27 Jun 2011 11:57:05 -0400
Subject: [SR-Users] LCR module
In-Reply-To: <SNT110-W598AA8AF2312A4A301DB7BCD570@phx.gbl>
References: <SNT110-W598AA8AF2312A4A301DB7BCD570@phx.gbl>
Message-ID: <SNT110-W52F1E2FDFB75EDB73654BCCD570@phx.gbl>


my bad, I forgot to look inside modules folder, it is there:

[root at registrar modules]# ls
auth.so    cfg_rpc.so       db_mysql.so  malloc_test.so  mqueue.so     prefix_route.so  sl.so        topoh.so
avpops.so  counters.so      debugger.so  matrix.so       mtree.so      privacy.so       sms.so       xhttp.so
blst.so    ctl.so           enum.so      mediaproxy.so   pdb.so        ratelimit.so     textopsx.so
cfg_db.so  db_flatstore.so  lcr.so       mi_rpc.so       pipelimit.so  sanity.so        tm.so
[root at registrar modules]#

also, I forgot to declare the default values of the AVPs

# ----- mi_fifo params -----
modparam("lcr", "gw_uri_avp", "$avp(i:709)")
modparam("lcr", "ruri_user_avp", "$avp(i:500)")
modparam("lcr", "flags_avp", "$avp(i:712)")
modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)")
modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)")

now, it loads fine

disregard this post


From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Date: Mon, 27 Jun 2011 11:45:32 -0400
Subject: [SR-Users] LCR module











After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time:



[root at registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/
[root at registrar modules_k]# ls
acc.so            db_text.so       drouting.so  mi_datagram.so    permissions.so  rtimer.so     statistics.so     usrloc.so
alias_db.so       db_unixodbc.so   exec.so      mi_fifo.so        pike.so         rtpproxy.so   textops.so        xlog.so
auth_db.so        dialog.so        group.so     msilo.so          pua_mi.so       siptrace.so   tmx.so
auth_diameter.so  dispatcher.so    htable.so    nathelper.so      pv.so           siputils.so   uac_redirect.so
benchmark.so      diversion.so     imc.so       nat_traversal.so  qos.so          speeddial.so  uac.so
call_control.so   domainpolicy.so  kex.so       path.so           registrar.so    sqlops.so     uri_db.so
cfgutils.so       domain.so        maxfwd.so    pdt.so            rr.so           sst.so        userblacklist.so

[root at registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/
[root at registrar modules_s]# ls
acc_db.so      avp.so         diversion.so  fifo.so    nathelper.so    pike.so       speeddial.so  uri_db.so
acc_syslog.so  db_ops.so      domain.so     gflags.so  options.so      print.so      textops.so    uri.so
auth_db.so     dialog.so      eval.so       maxfwd.so  pdt.so          registrar.so  timer.so      usrloc.so
avp_db.so      dispatcher.so  exec.so       msilo.so   permissions.so  rr.so         uac.so        xlog.so
[root at registrar modules_s]#

I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something?

This is from the syslog

Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume
Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume

and this is from the fifo interface:

[root at registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload
500 command 'lcr' not available
[root at registrar kamailio]#

 		 	   		  

_______________________________________________
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 alhakeem at gmail.com  Mon Jun 27 19:03:01 2011
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Mon, 27 Jun 2011 18:03:01 +0100
Subject: [SR-Users] Proxy setup in a non-recursive mode
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEEwH9YGNtFIts3Q9/Tg/3YBAAAAAA==@gmail.com>

 
Hello,
Could anyone help with suggestions on how to setup a Proxy ( R-Proxy) that only
does a route lookup and pass the parameter back to origination Proxy/Gateway A
to forward the call based on the new parameter.
 
On receipt of an invite from GW/Proxy A,   R- Proxy will do a route lookup for
the called number 447988870290,
If it finds a route then it should append the network prefix to the new INVITE
or rewrite the INVITE 23430442088870290 in the 302 Moved Temporarily message
sent back to GW/Proxy A in the Contact field and there are two CC-Diversion
header fields included; at GW/Proxy (IP address) who originated the call and
already has a route or dial plan for the new prefix and also another one for
same at GW/Proxy (IP address).
 
Is there any way to do this without creating a loop in the network ?
 
Previously this setup can be achieved with the originating network having 2
GW/Proxies  i.e. GW/Proxy A  and GW/Proxy B. In the 302 Temporarily moved
message  the Contact will show GW/Proxy B, while the 2 CC-Diversion header
fields will add GW/Proxy B  and GW Proxy A. 
 
Regards,
Abdul Hakeem
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From miconda at gmail.com  Mon Jun 27 19:09:33 2011
From: miconda at gmail.com (Daniel-Constantine Mierla)
Date: Mon, 27 Jun 2011 19:09:33 +0200
Subject: [SR-Users] [sr-dev] new module: async - asynchronous SIP
	request processing in cfg
In-Reply-To: <870716D7-124A-4F66-B770-D523BDE6B7E6@edvina.net>
References: <4E083FFF.6020902@gmail.com>
	<870716D7-124A-4F66-B770-D523BDE6B7E6@edvina.net>
Message-ID: <7DC833D8-800C-452E-989E-E5A271D13E1D@gmail.com>



On Jun 27, 2011, at 4:11 PM, Olle E. Johansson <oej at edvina.net> wrote:

> 
> 27 jun 2011 kl. 10.31 skrev Daniel-Constantin Mierla:
> 
>> Hello,
>> 
>> a new module is now in GIT master branch, named async.
>> 
>> Its purpose is to provide asynchronous SIP request processing using t_suspend()/t_continue() mechanism from tm module. While the tm module had it for quite some time, it was not available for usage in configuration file.
>> 
>> There are two functions implemented by now:
>> 
>> - async_route(routename, interval) - execute asynchronously a route block after a time interval
>> - aync_sleep(interval) - sleep asynchronously for a time interval and then resume SIP request processing
>> 
>> See the readme for more details:
>> http://kamailio.org/docs/modules/devel/modules/async.html
>> 
> Seems kind of sexy.
> 
> Do you have a good example of usage?

> 
Delayed forwarding of requests or delayed replying - like 100 trying, 180 ringing immediately when invite comes in, then async_route() to reply 408 timeout after 20 sec when you run out of gw capacity :-).

All without blocking main sip workers.

For SIP MESSAGE to SMS you can use the mechanism just to switch to other pool of workers in case SMS gatewaying is blocking - although this can be achieved as well with mqueue and rtimer modules if you don't need the full sip request in special workers.

Just some thoughts ...

Hopefully the module will help others to contribute more functions that will work asynchronously. My todo has some more ideas here as well, just that is as long as Madrid - Berlin, so may not get in that soon :-)

Enjoy the cold Saharian daylight temperatures,
Daniel

From fborot at hotmail.com  Mon Jun 27 23:23:05 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Mon, 27 Jun 2011 17:23:05 -0400
Subject: [SR-Users] Questions about LCR module
Message-ID: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>



I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module.
I have these main questions:

1- does lcr_id replaces the grp_id?
2- what happened to the fifo commands to add gws etc:

 lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip>
           ............... add a gateway with flags, tag and strip ............
           ................(flags, tag, and strip are optional arguments) .....
 lcr rmgw  <gw_name> ..........................................................
           ............... delete a gateway
 lcr addroute <prefix> <from> <grp_id> <prio> .................................
           .............. add a route ( use '' to match anything in <from> )
 lcr rmroute  <prefix> <from> <grp_id> <prio> .................................
           .............. delete a route


3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload"  only the lcr tables

What would be the best way to use this module on this version?

Any help/hint would be greatly appreciated.
thank you

fborot





  		 	   		  
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From gminghon at gmail.com  Tue Jun 28 04:25:01 2011
From: gminghon at gmail.com (MingHon)
Date: Tue, 28 Jun 2011 10:25:01 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTimVwSAv9JZBUZj5N3kPbcs-KXE-Nw@mail.gmail.com>

Hi,

i registered 3 uac behind same nat successfully but when i try to call each
other i didnt get any audio.
but if i use uac 102 and 103 to call into the voicemail i heard the audio
but not for 101.
kamailio is listening 60.48.218.61 and 192.168.2.3
rtpproxy is running.
asterisk is at 192.168.2.23.

here is my ul show.

AOR:: 102
                Contact:: sip:102 at 175.136.221.60:5062 Q=
                        Expires:: 3110
                        Callid:: 721498432 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: T22 7.3.0.50
                        Received:: sip:175.136.221.60:1024
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 192
                        Socket:: udp:60.48.218.61:5060
                        Methods:: 16383
        AOR:: 103
                Contact:: sip:103 at 175.136.221.60:5062 Q=
                        Expires:: 3114
                        Callid:: 1499738216 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: Yealink SIP-T18 18.0.0.70
                        Received:: sip:175.136.221.60:1025
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 192
                        Socket:: udp:60.48.218.61:5060
                        Methods:: 16383
        AOR:: 101
                Contact:: sip:101 at 175.136.221.60:5062 Q=
                        Expires:: 3097
                        Callid:: 166053301 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: T20 9.41.0.80
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 0
                        Socket:: udp:60.48.218.61:5060
                        Methods:: 16383

and may i know why uac 101 did not have the received: field?

please some one could give a hand on this? the audio really cant get thru i
really have no idea.

thank you

-- 
Regards,

MingHon
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From gminghon at gmail.com  Tue Jun 28 07:53:57 2011
From: gminghon at gmail.com (MingHon)
Date: Tue, 28 Jun 2011 13:53:57 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTimVAq=Dsuxptm_+XqNo5zaOKA8vJQ@mail.gmail.com>

Hi,

i fixed the audio issue for 102 to 103 vice versa.

by fixing the canreinvite in asterisk.

from uac the rtp packet will route to kamailio den forward to asterisk.

can we bypass the rtp packet going to asterisk?

and here is the update for uac 101 issue.

when 101 call to voicemail or 102/103 there is no audio.

in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.

but if 102/103 call to 101 both uac got audio.

i realize this is because 101 is the first uac registered before 102/103 and
because it did not have the received: field in ul show.

please adv.

-- 
Regards,

MingHon
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From solanki.chandrakant at gmail.com  Tue Jun 28 07:57:41 2011
From: solanki.chandrakant at gmail.com (Chandrakant Solanki)
Date: Tue, 28 Jun 2011 11:27:41 +0530
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTimVAq=Dsuxptm_+XqNo5zaOKA8vJQ@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
	<BANLkTimVAq=Dsuxptm_+XqNo5zaOKA8vJQ@mail.gmail.com>
Message-ID: <BANLkTikpxFfDDGzLCKujaa=WQZO4yPGiyg@mail.gmail.com>

Try to run rtpproxy on private ip not on local 127.0.0.1

On Tue, Jun 28, 2011 at 11:23 AM, MingHon <gminghon at gmail.com> wrote:

> Hi,
>
> i fixed the audio issue for 102 to 103 vice versa.
>
> by fixing the canreinvite in asterisk.
>
> from uac the rtp packet will route to kamailio den forward to asterisk.
>
> can we bypass the rtp packet going to asterisk?
>
> and here is the update for uac 101 issue.
>
> when 101 call to voicemail or 102/103 there is no audio.
>
> in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.
>
> but if 102/103 call to 101 both uac got audio.
>
> i realize this is because 101 is the first uac registered before 102/103
> and because it did not have the received: field in ul show.
>
> please adv.
>
> --
> Regards,
>
> MingHon
>
> _______________________________________________
> 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
>
>


-- 
Regards,

Chandrakant Solanki
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From gminghon at gmail.com  Tue Jun 28 08:13:48 2011
From: gminghon at gmail.com (MingHon)
Date: Tue, 28 Jun 2011 14:13:48 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTindip7qFkbwZw3=Y08aQbQ_bonuwA@mail.gmail.com>

Hi,

Thanks for the suggestion. i tried running rtpproxy on private.

rtpproxy -l public_ip -s udp:192.168.2.3:7722 -u user

and in kamailio cfg.

modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.2.3:7722")

but still the same issue..

thanks.

-- 
Regards,

MingHon
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From klaus.mailinglists at pernau.at  Tue Jun 28 09:45:52 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 28 Jun 2011 09:45:52 +0200
Subject: [SR-Users] Proxy setup in a non-recursive mode
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEEwH9YGNtFIts3Q9/Tg/3YBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEEwH9YGNtFIts3Q9/Tg/3YBAAAAAA==@gmail.com>
Message-ID: <4E0986B0.5010304@pernau.at>

Being the "R-Proxy" is quite easy with Kamailio.

# lookup route and set new RURI
...
prefix("23430");

# add any custom headers to response
append_to_reply(.....);


# send 302 (new RURI will automatically added as Contact header)
sl_send_reply("302","Moved");
exit;


Of course Kamailio has no influence how the Gateway will behave on
reception of the 302 response.

regards
Klaus


Am 27.06.2011 19:03, schrieb Abdul Hakeem:
>  
> 
> Hello,
> 
> Could anyone help with suggestions on how to setup a Proxy ( R-Proxy)
> that only does a route lookup and pass the parameter back to origination
> Proxy/Gateway A to forward the call based on the new parameter.
> 
>  
> 
> On receipt of an invite from GW/Proxy A,   R- Proxy will do a route
> lookup for the called number 447988870290,
> 
> If it finds a route then it should append the network prefix to the new
> INVITE or rewrite the INVITE 23430442088870290 in the 302 Moved
> Temporarily message sent back to GW/Proxy A in the Contact field and
> there are two CC-Diversion header fields included; at GW/Proxy (IP
> address) who originated the call and already has a route or dial plan
> for the new prefix and also another one for same at GW/Proxy (IP address).
> 
>  
> 
> Is there any way to do this without creating a loop in the network ?
> 
>  
> 
> Previously this setup can be achieved with the originating network
> having 2 GW/Proxies  i.e. GW/Proxy A  and GW/Proxy B. In the 302
> Temporarily moved message  the Contact will show GW/Proxy B, while the 2
> CC-Diversion header fields will add GW/Proxy B  and GW Proxy A.
> 
>  
> 
> Regards,
> 
> Abdul Hakeem
> 
> 
> 
> _______________________________________________
> 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  Tue Jun 28 10:02:20 2011
From: phillman25 at gmail.com (Phillman25 Kyriacou)
Date: Tue, 28 Jun 2011 11:02:20 +0300
Subject: [SR-Users] Kamailio and DTMF
Message-ID: <BANLkTi=DeFAtwV+gcyP1e9XwNYWR_e9k8g@mail.gmail.com>

Hi

I was wondering what DTMF modes Kamailio supports? If i wanted to force a
DTMF mode on Kamailio how could i go about doing it?

thanking you in advance

Phillip
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From gminghon at gmail.com  Tue Jun 28 10:03:58 2011
From: gminghon at gmail.com (MingHon)
Date: Tue, 28 Jun 2011 16:03:58 +0800
Subject: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
In-Reply-To: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
References: <BANLkTin2WsZ3VZNSsnd+Khw_RsUobi2WVA@mail.gmail.com>
Message-ID: <BANLkTi=qkCFq-UMK9Pk-Ln0gMt2Vmz5oUw@mail.gmail.com>

Hi List,

i tcpdump on my kamailio server.

when uac registered with source port 5060

the uac will remain  Cflag:: 0 and did not have the Received: field.

i also tried some iphone sip apps.

some uac app will register with src port 5060 and some will not.

anyone have solution for this?

or how to prevent uac register with src port 5060 ?

thanks!

-- 
Regards,

MingHon
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From andrew.pogrebennyk at portaone.com  Tue Jun 28 10:07:16 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Tue, 28 Jun 2011 11:07:16 +0300
Subject: [SR-Users] Kamailio and DTMF
In-Reply-To: <BANLkTi=DeFAtwV+gcyP1e9XwNYWR_e9k8g@mail.gmail.com>
References: <BANLkTi=DeFAtwV+gcyP1e9XwNYWR_e9k8g@mail.gmail.com>
Message-ID: <4E098BB4.1000709@portaone.com>

Phillip,
This question is irrelevant to Kamailio. The DTMF mode and codecs are 
negotiated between the two endpoints only, so Kamailio supports any codec.

On 28.06.2011 11:06, Phillman25 Kyriacou wrote:
> Hi
> 
> I was wondering what DTMF modes Kamailio supports? If i wanted to force 
> a DTMF mode on Kamailio how could i go about doing it?
> 
> thanking you in advance
> 
> Phillip

-- 
Sincerely,
Andrew Pogrebennyk


From rabs at dimension-virtual.com  Tue Jun 28 10:16:06 2011
From: rabs at dimension-virtual.com (=?iso-8859-1?B?UmHDumw=?= Alexis Betancor Santana)
Date: Tue, 28 Jun 2011 09:16:06 +0100
Subject: [SR-Users] Kamailio and DTMF
In-Reply-To: <4E098BB4.1000709@portaone.com>
References: <BANLkTi=DeFAtwV+gcyP1e9XwNYWR_e9k8g@mail.gmail.com>
	<4E098BB4.1000709@portaone.com>
Message-ID: <20110628081605.GA8769@salma>

On Tue, Jun 28, 2011 at 11:07:16AM +0300, Andrew Pogrebennyk wrote:
> Phillip,
> This question is irrelevant to Kamailio. The DTMF mode and codecs
> are negotiated between the two endpoints only, so Kamailio supports
> any codec.


This is not fully true on the DTMF side, if you use SIP-INFO as DMTF
transport, DTMF's will go throught kamailio (if configured to do so).

Best regards


From andrew.pogrebennyk at portaone.com  Tue Jun 28 10:19:20 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Tue, 28 Jun 2011 11:19:20 +0300
Subject: [SR-Users] Kamailio and DTMF
In-Reply-To: <20110628081605.GA8769@salma>
References: <BANLkTi=DeFAtwV+gcyP1e9XwNYWR_e9k8g@mail.gmail.com>	<4E098BB4.1000709@portaone.com>
	<20110628081605.GA8769@salma>
Message-ID: <4E098E88.6030502@portaone.com>

On 28.06.2011 11:18, Ra?l Alexis Betancor Santana wrote:
> This is not fully true on the DTMF side, if you use SIP-INFO as DMTF
> transport, DTMF's will go throught kamailio (if configured to do so).

Right - but then this is achieved through the configuration script.

Andrew


From keeling at akan-tech.com  Tue Jun 28 10:37:11 2011
From: keeling at akan-tech.com (Nathaniel L Keeling)
Date: Tue, 28 Jun 2011 03:37:11 -0500
Subject: [SR-Users] Kamailio 64bit Compile Errors
Message-ID: <4E0992B7.5050602@akan-tech.com>

I am trying to install Kamailio 3.1.0 on Solaris 10 that is a sparc 
machine (v120) and is running a 64bit OS. When I originally performed 
the compile utilizing the instructions from the website, I was getting 
warnings about using 32 bit headers. I was able to compile and install 
successfully, but when I started Kamailio, loading of modules were 
failing due to "not found" error. The warning messages:

  "atomic/atomic_sparc.h:41:2: warning: #warning "sparc32 atomic 
operations support not tested""

I thought was because the target architecture specified sparc and not 
sparc64, so I tried specified the target architecture as listed below 
which got rid of the warning messages but then I got ELFCLASS32 error 
messages. What is the proper way to create a 64bit version of Kamailio 
on a sparc Solaris 10. It seems like the target architecure is not being 
configured properly and that may be causing other problems. I have 
included output from different commands about the environment and the 
commands that I entered to compile Kamailio.

# make FLAVOUR=kamailio cfg
target architecture <sparc>, host architecture <sparc64>

# isainfo -n
sparcv9
# uname -p
sparc
# uname -m
sun4u
# isainfo -v
64-bit sparcv9 applications
         vis
32-bit sparc applications
         vis v8plus div32 mul32
# isainfo -b
64

Here is what I entered to try and force the building of a 64 bit version 
and a sample error that I was getting.

make FLAVOUR=kamailio prefix=/usr/local/kamailio-3.1 CPU=ultrasparc 
ARCH=sparc64 cfg



LD (gcc) [M lcr.so]             lcr.so
ld: fatal: file /usr/local/lib/libpcre.so: wrong ELF class: ELFCLASS32
ld: fatal: File processing errors. No output written to lcr.so
collect2: ld returned 1 exit status
make[1]: *** [lcr.so] Error 1
make: *** [modules] Error 1

LD_LIBRARY_PATH=/lib/sparcv9:/usr/lib/sparcv9:/usr/sfw/lib/sparcv9:/usr/local/postgres/9.0-pgdg/lib/64
LD_RUN_PATH=/lib/sparcv9:/usr/lib/sparcv9:/usr/sfw/lib/sparcv9:/usr/local/postgres/9.0-pgdg/lib/64

PATH=/opt/csw/bin:/usr/sfw/bin:/usr/local/bin:/usr/local/postgres/9.0-pgdg/bin:/usr/sbin:/usr/bin:/usr/ccs/bin:

# find . -name libpcre.so -print
./usr/local/lib/libpcre.so


Thanks



From danb.lists at googlemail.com  Tue Jun 28 11:58:43 2011
From: danb.lists at googlemail.com (Dan-Cristian Bogos)
Date: Tue, 28 Jun 2011 11:58:43 +0200
Subject: [SR-Users] [sr-dev] new module: async - asynchronous SIP
 request processing in cfg
Message-ID: <BANLkTingi9pg8s4rENEEWUj8L_Z0qV1yyA@mail.gmail.com>

Hey Daniel,

As said in the past, this module is good news, thanks for making it available.
As a big fan of delayed billing, sleep() does a great job of delaying
200 OK from the upstream for me (useful for example with analogue/CDMA
gateways sending 200 OK on ringback).

Cheers,
DanB


From jh at tutpro.com  Tue Jun 28 12:14:13 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Tue, 28 Jun 2011 13:14:13 +0300
Subject: [SR-Users]  Questions about LCR module
In-Reply-To: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>
References: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>
Message-ID: <19977.43381.690535.657546@sip.test.fi>

Fabian Borot writes:

> I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see
> huge differences on the LCR module.

there is a script somewhere (read wiki) that upgrades the tables.

> I have these main questions:
> 
> 1- does lcr_id replaces the grp_id?

not really.  lcr_ids denote independent lcr instances.

> 2- what happened to the fifo commands to add gws etc:

fifo commands to manipulate lcr tables are not supported by me.

-- juha


From jgallartm at gmail.com  Tue Jun 28 17:53:41 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 28 Jun 2011 17:53:41 +0200
Subject: [SR-Users] Dialog module question
Message-ID: <BANLkTim1aPhxidyBNF6iV22jdtn0TQNN4w@mail.gmail.com>

Hello

I have little experience with the dialog module -so sorry if the quesiton is
silly-, and I was wondering if I could use it for doing accounting (acc
works just fine, but I was looking for a unique stop record that includes
all the call information). The basic idea would be to assign the initial
INVITE to a profile; and then, when BYE is received, build a cdr based of
the dialog attributes...but afaik , upon reception of the BYE the dialog is
destroyed. is there a way to access the dlg attributes inside the script
when the BYE transaction is processed?

Thanks in advance

Javi
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From scramatte at gmail.com  Tue Jun 28 18:19:06 2011
From: scramatte at gmail.com (=?ISO-8859-1?Q?S=E9bastien_Cramatte?=)
Date: Tue, 28 Jun 2011 18:19:06 +0200
Subject: [SR-Users] Kamailio+ Asterisk 491 error code ????
Message-ID: <BANLkTi=nEQok296KQDA+LYg32h-82kiR8w@mail.gmail.com>

Hello,

I've just setup Kamailio+ Asterisk following   Daniel  Constantin Mierla
howto

http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb


Everything seemts to works and  I'm able to place a call to local
extensions.
I'm able to establish a call but strangely  I get and   491 error code  on
my  SIP Phones

In my Asterisk console I get this error :

[Jun 28 18:14:42] WARNING[24432]: chan_sip.c:18087 handle_response_invite:
just did sched_add waitid(23) for sip_reinvite_retry for dialog
LIFuQLu1l3ZtKrNy0XDX7lEk.Y5JdFko in handle_response_invite
  == Spawn extension (from-local, 102, 1) exited non-zero on
'SIP/103-00000000'


Anybody can give me an hand to solve this issue ?


Best regards
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From lau.testi at gmail.com  Tue Jun 28 18:33:40 2011
From: lau.testi at gmail.com (laura testi)
Date: Tue, 28 Jun 2011 18:33:40 +0200
Subject: [SR-Users] =?windows-1252?q?xcap_server_-_http_load=96balancer=3A?=
	=?windows-1252?q?_xcap_authentication_problem?=
Message-ID: <BANLkTim2D87vqzFy59OQ7iTeORMDrosAhg@mail.gmail.com>

Hello all,

I?m using two Kamailio Presence Servers that are also XCAP Servers for the
presence and resources management and the Kamailio Dispatcher in order to
balance the SIP requestes coming from SIP Client.

Then I?m trying to configure a HTTP  load?balancer for the xcap/http
traffic.

I?m testing both Apache and INginx HTTP proxy but I?m facing with a problem
related with the Xcap authentication:

when a client sends the first xcap request to the http load-balancer, it
forwards the request to one of the configured Xcap Server that replies with
the 401 response in order to let the client authenticates. So, the client
sends the second request with the Digest authentication to the http
load-balancer. But, sometimes, the http load-balancer forwards the
authenticate request to the second Xcap Server that is not able to
authenticate the request.  Then the client receive the ?Unauthorized?
response?



Could you please give us some hints about this problem?

Thanks in advance,

laura
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From timo.reimann at 1und1.de  Tue Jun 28 19:34:47 2011
From: timo.reimann at 1und1.de (Timo Reimann)
Date: Tue, 28 Jun 2011 19:34:47 +0200
Subject: [SR-Users] Dialog module question
In-Reply-To: <BANLkTim1aPhxidyBNF6iV22jdtn0TQNN4w@mail.gmail.com>
References: <BANLkTim1aPhxidyBNF6iV22jdtn0TQNN4w@mail.gmail.com>
Message-ID: <4E0A10B7.9010009@1und1.de>

Hey Javier,


On 28.06.2011 17:53, Javier Gallart wrote:
> I have little experience with the dialog module -so sorry if the
> quesiton is silly-, and I was wondering if I could use it for doing
> accounting (acc works just fine, but I was looking for a unique stop
> record that includes all the call information). The basic idea would be
> to assign the initial INVITE to a profile; and then, when BYE is
> received, build a cdr based of the dialog attributes...but afaik , upon
> reception of the BYE the dialog is destroyed. is there a way to access
> the dlg attributes inside the script when the BYE transaction is processed?

Basically, what you are referring to is dialog-based CDR generation.
Although it's not possible to do in the current Kamailio, a co-worker of
mine and me have been working hard lately to implement this specific
feature. It requires changes to the dialog module (such as allowing
access to dialogs even after the BYE message is processed, just as you
mentioned) and extending the acc module by a CDR mode.

The plan is to get the new code into the project ASAP, certainly by the
time SR 3.2 is released. So watch out for a new repository branch and an
announcement on the mailing list to get your hands dirty with the new
stuff soon.


Cheers,

--Timo


From miconda at gmail.com  Tue Jun 28 20:43:04 2011
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 28 Jun 2011 20:43:04 +0200
Subject: [SR-Users]
 =?windows-1252?q?xcap_server_-_http_load=96balancer=3A?=
 =?windows-1252?q?_xcap_authentication_problem?=
In-Reply-To: <BANLkTim2D87vqzFy59OQ7iTeORMDrosAhg@mail.gmail.com>
References: <BANLkTim2D87vqzFy59OQ7iTeORMDrosAhg@mail.gmail.com>
Message-ID: <4E0A20B8.1090203@gmail.com>

Hello,


On 6/28/11 6:33 PM, laura testi wrote:
>
> Hello all,
>
> I?m using two Kamailio Presence Servers that are also XCAP Servers for 
> the presence and resources management and the Kamailio Dispatcher in 
> order to balance the SIP requestes coming from SIP Client.
>
> Then I?m trying to configure a HTTP load?balancer for the xcap/http 
> traffic.
>
> I?m testing both Apache and INginx HTTP proxy but I?m facing with a 
> problem related with the Xcap authentication:
>
> when a client sends the first xcap request to the http load-balancer, 
> it forwards the request to one of the configured Xcap Server that 
> replies with the 401 response in order to let the client 
> authenticates. So, the client sends the second request with the Digest 
> authentication to the http load-balancer. But, sometimes, the http 
> load-balancer forwards the authenticate request to the second Xcap 
> Server that is not able to authenticate the request. Then the client 
> receive the ?Unauthorized? response?
>
> Could you please give us some hints about this problem?
>

you have to set the same value to secret parameter of auth module:
http://kamailio.org/docs/modules/stable/modules/auth.html#auth.secret

Also be sure the two servers have the same time (use ntp or so) and the 
one_time_nonce parameter to auth module is off (not enabled).

Then it should work.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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From jgallartm at gmail.com  Tue Jun 28 21:52:44 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 28 Jun 2011 21:52:44 +0200
Subject: [SR-Users] Dialog module question
In-Reply-To: <4E0A10B7.9010009@1und1.de>
References: <BANLkTim1aPhxidyBNF6iV22jdtn0TQNN4w@mail.gmail.com>
	<4E0A10B7.9010009@1und1.de>
Message-ID: <BANLkTimAcejsn2uT6q+1pYWnA6RKOmjSAg@mail.gmail.com>

Thanks Timo

it looks really promising! I'll stay tuned for upcoming annoucements...

Regards

Javi

On Tue, Jun 28, 2011 at 7:34 PM, Timo Reimann <timo.reimann at 1und1.de> wrote:

> Hey Javier,
>
>
> On 28.06.2011 17:53, Javier Gallart wrote:
> > I have little experience with the dialog module -so sorry if the
> > quesiton is silly-, and I was wondering if I could use it for doing
> > accounting (acc works just fine, but I was looking for a unique stop
> > record that includes all the call information). The basic idea would be
> > to assign the initial INVITE to a profile; and then, when BYE is
> > received, build a cdr based of the dialog attributes...but afaik , upon
> > reception of the BYE the dialog is destroyed. is there a way to access
> > the dlg attributes inside the script when the BYE transaction is
> processed?
>
> Basically, what you are referring to is dialog-based CDR generation.
> Although it's not possible to do in the current Kamailio, a co-worker of
> mine and me have been working hard lately to implement this specific
> feature. It requires changes to the dialog module (such as allowing
> access to dialogs even after the BYE message is processed, just as you
> mentioned) and extending the acc module by a CDR mode.
>
> The plan is to get the new code into the project ASAP, certainly by the
> time SR 3.2 is released. So watch out for a new repository branch and an
> announcement on the mailing list to get your hands dirty with the new
> stuff soon.
>
>
> Cheers,
>
> --Timo
>
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From fborot at hotmail.com  Tue Jun 28 22:48:49 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Tue, 28 Jun 2011 16:48:49 -0400
Subject: [SR-Users] Questions about LCR module
In-Reply-To: <SNT110-W571E3FE9842B1B21139700CD560@phx.gbl>
References: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>,
	<SNT110-W61D8ABC08C48B487A15EFACD560@phx.gbl>,
	<SNT110-W571E3FE9842B1B21139700CD560@phx.gbl>
Message-ID: <SNT110-W11CE1518BA4DE438B67B63CD560@phx.gbl>
























Thanks Juha
I populated the lcr tables by hand. This is how they look now, I have 2 lcr_ids [1 and 2], each instance has its own gw. And each instance has its own prefix to match for [4444 and 5555].
The idea is that a call comes in with R-URI 444413051234567, it should match the lcr_id= 200, then it should go out using gw 192.168.169.200 etc

mysql> select * from lcr_gw;
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
| id | lcr_id | gw_name    | ip_addr         | hostname          | port | params | uri_scheme | transport | strip | tag  | flags | defunct |
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
|  1 |    200 | switchvox  | 192.168.169.200 | pbx1.mydomain.com | 5060 | none   |          1 |         1 |     0 |      |     0 |       0 |
|  2 |    201 | switchvox2 | 192.168.169.201 | pbx1.mydomain.net | 5060 | none   |          1 |         1 |     0 |      |     0 |       0 |
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
mysql> select * from lcr_rule;
+----+--------+--------+----------+---------+---------+
| id | lcr_id | prefix | from_uri | stopper | enabled |
+----+--------+--------+----------+---------+---------+
|  1 |    200 | 4444   |          |       0 |       1 |
|  2 |    201 | 5555   |          |       0 |       1 |
+----+--------+--------+----------+---------+---------+
mysql> select * from lcr_rule_target;
+----+--------+---------+-------+----------+--------+
| id | lcr_id | rule_id | gw_id | priority | weight |
+----+--------+---------+-------+----------+--------+
|  1 |    200 |       1 |     1 |        5 |      1 |
|  2 |    201 |       2 |     2 |        5 |      1 |
+----+--------+---------+-------+----------+--------+

Then I tried the sercmd commands to reload and dump and I do not see the contents o the screen. Is this how is supposed to work now?. How do I know that the new values are being loaded into memory w/o restarting kamailio?

I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see
> huge differences on the LCR module.

there is a script somewhere (read wiki) that upgrades the tables.

> I have these main questions:
> 
> 1- does lcr_id replaces the grp_id?

not really.  lcr_ids denote independent lcr instances.

> 2- what happened to the fifo commands to add gws etc:

fifo commands to manipulate lcr tables are not supported by me.

-- juha



 

From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Subject: Questions about LCR module
Date: Mon, 27 Jun 2011 17:23:05 -0400









I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module.
I have these main questions:

1- does lcr_id replaces the grp_id?
2- what happened to the fifo commands to add gws etc:

 lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip>
           ............... add a gateway with flags, tag and strip ............
           ................(flags, tag, and strip are optional arguments) .....
 lcr rmgw  <gw_name> ..........................................................
           ............... delete a gateway
 lcr addroute <prefix> <from> <grp_id> <prio> .................................
           .............. add a route ( use '' to match anything in <from> )
 lcr rmroute  <prefix> <from> <grp_id> <prio> .................................
           .............. delete a route


3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload"  only the lcr tables

What would be the best way to use this module on this version?

Any help/hint would be greatly appreciated.
thank you

fborot





  		 	   		   		 	   		   		 	   		   		 	   		  
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From klaus.mailinglists at pernau.at  Tue Jun 28 22:58:03 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Tue, 28 Jun 2011 22:58:03 +0200
Subject: [SR-Users] Kamailio+ Asterisk 491 error code ????
In-Reply-To: <BANLkTi=nEQok296KQDA+LYg32h-82kiR8w@mail.gmail.com>
References: <BANLkTi=nEQok296KQDA+LYg32h-82kiR8w@mail.gmail.com>
Message-ID: <4E0A405B.5070301@pernau.at>

S?bastien Cramatte wrote:
> Hello,
> 
> I've just setup Kamailio+ Asterisk following   Daniel  Constantin 
> Mierla  howto
> 
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
> 
> 
> Everything seemts to works and  I'm able to place a call to local 
> extensions.
> I'm able to establish a call but strangely  I get and   491 error code  
> on my  SIP Phones
> 
> In my Asterisk console I get this error :
> 
> [Jun 28 18:14:42] WARNING[24432]: chan_sip.c:18087 
> handle_response_invite: just did sched_add waitid(23) for 
> sip_reinvite_retry for dialog LIFuQLu1l3ZtKrNy0XDX7lEk.Y5JdFko in 
> handle_response_invite
>   == Spawn extension (from-local, 102, 1) exited non-zero on 
> 'SIP/103-00000000'

For some reason Asterisk hang up the channel. Enable verbose logging and 
watch how Asterisk processes your dialplan (in the context from-local).

If you are stuck, ask on the asterisk-users mailing list.

regards
Klaus


From Bradley.Watkins at compuware.com  Tue Jun 28 23:36:33 2011
From: Bradley.Watkins at compuware.com (Watkins, Bradley)
Date: Tue, 28 Jun 2011 21:36:33 +0000
Subject: [SR-Users] mq_fetch and while loops
Message-ID: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>

I'm having an issue trying to use the construct show in the module documentation.

The docs show:
while(mq_fetch("myq"))
{
   xlog("$mqk(myq) - $mqv(myq)\n");
}

And I'm doing essentially the same thing (differently named mqueue, but otherwise identical), but invariably I get this in the logs:
ERROR: <core> [action.c:1388]: ERROR: runaway while (559, 28): more then 100 loops

Am I doing something wrong, or is mq_fetch not returning appropriate values for successful versus failed fetches?
This is on Kamailio 3.1.4, FWIW.

Regards,
- Brad

The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.



From abalashov at evaristesys.com  Tue Jun 28 23:41:40 2011
From: abalashov at evaristesys.com (Alex Balashov)
Date: Tue, 28 Jun 2011 17:41:40 -0400
Subject: [SR-Users] mq_fetch and while loops
In-Reply-To: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>
References: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>
Message-ID: <4E0A4A94.8090706@evaristesys.com>

On 06/28/2011 05:36 PM, Watkins, Bradley wrote:

> I'm having an issue trying to use the construct show in the module documentation.
>
> The docs show:
> while(mq_fetch("myq"))
> {
>     xlog("$mqk(myq) - $mqv(myq)\n");
> }
>
> And I'm doing essentially the same thing (differently named mqueue, but otherwise identical), but invariably I get this in the logs:
> ERROR:<core>  [action.c:1388]: ERROR: runaway while (559, 28): more then 100 loops

I reported this back in October:

    http://lists.sip-router.org/pipermail/sr-users/2010-October/065740.html

A fix was committed to the master branch almost immediately:

    http://lists.sip-router.org/pipermail/sr-dev/2010-October/009499.html

I requested that it be backported into the 3.1 branch in May and it was 
done:

    http://lists.sip-router.org/pipermail/sr-dev/2011-May/011132.html

However, I don't know that it's made it into the tarballs yet.  You can 
get the 3.1 branch with the fix by doing this:

    git clone git://git.sip-router.org/sip-router -b 3.1

Otherwise, here's the workaround I was using before:

       while(mq_fetch("myq")) {
          ...

          if(!defined $mqk(myq))
             break;

           ...
        }

Cheers,

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/


From Bradley.Watkins at compuware.com  Tue Jun 28 23:53:45 2011
From: Bradley.Watkins at compuware.com (Watkins, Bradley)
Date: Tue, 28 Jun 2011 21:53:45 +0000
Subject: [SR-Users] mq_fetch and while loops
In-Reply-To: <4E0A4A94.8090706@evaristesys.com>
References: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>
	<4E0A4A94.8090706@evaristesys.com>
Message-ID: <C778BE90B341194A960B1EF37812EA6A0C3C9888@nasa-dtw-mbx02.nasa.cpwr.corp>

>
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

>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, June 28, 2011 5:42 PM
>To: sr-users at lists.sip-router.org
>Subject: Re: [SR-Users] mq_fetch and while loops
>
>I reported this back in October:
>
>    http://lists.sip-router.org/pipermail/sr-users/2010-October/065740.html
>
>A fix was committed to the master branch almost immediately:
>
>    http://lists.sip-router.org/pipermail/sr-dev/2010-October/009499.html
>
>I requested that it be backported into the 3.1 branch in May and it was
>done:
>
>    http://lists.sip-router.org/pipermail/sr-dev/2011-May/011132.html
>
>However, I don't know that it's made it into the tarballs yet.  You can get the
>3.1 branch with the fix by doing this:
>
>    git clone git://git.sip-router.org/sip-router -b 3.1
>
>Otherwise, here's the workaround I was using before:
>
>       while(mq_fetch("myq")) {
>          ...
>
>          if(!defined $mqk(myq))
>             break;
>
>           ...
>        }
>
>Cheers,
>
>-- Alex

Excellent, thanks for the quick response!

Regards,
- Brad



From Bradley.Watkins at compuware.com  Wed Jun 29 00:18:35 2011
From: Bradley.Watkins at compuware.com (Watkins, Bradley)
Date: Tue, 28 Jun 2011 22:18:35 +0000
Subject: [SR-Users] mq_fetch and while loops
In-Reply-To: <C778BE90B341194A960B1EF37812EA6A0C3C9888@nasa-dtw-mbx02.nasa.cpwr.corp>
References: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>
	<4E0A4A94.8090706@evaristesys.com>
	<C778BE90B341194A960B1EF37812EA6A0C3C9888@nasa-dtw-mbx02.nasa.cpwr.corp>
Message-ID: <C778BE90B341194A960B1EF37812EA6A0C3C98D8@nasa-dtw-mbx02.nasa.cpwr.corp>

Just a correction to my previous statement:
Apparently I was incorrect and I was running Kamailio 3.1.3 and not 3.1.4.  I have installed 3.1.4 and it seems this fix is indeed in a released version now.

Regards,
- Brad

>
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

>From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-
>bounces at lists.sip-router.org] On Behalf Of Watkins, Bradley
>Sent: Tuesday, June 28, 2011 5:54 PM
>To: sr-users at lists.sip-router.org
>Subject: Re: [SR-Users] mq_fetch and while loops
you received it in error please notify us immediately and
>then destroy it.
>
>>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, June 28, 2011 5:42 PM
>>To: sr-users at lists.sip-router.org
>>Subject: Re: [SR-Users] mq_fetch and while loops
>>
>>I reported this back in October:
>>
>>
>> http://lists.sip-router.org/pipermail/sr-users/2010-October/065740.htm
>> l
>>
>>A fix was committed to the master branch almost immediately:
>>
>>
>> http://lists.sip-router.org/pipermail/sr-dev/2010-October/009499.html
>>
>>I requested that it be backported into the 3.1 branch in May and it was
>>done:
>>
>>    http://lists.sip-router.org/pipermail/sr-dev/2011-May/011132.html
>>
>>However, I don't know that it's made it into the tarballs yet.  You can
>>get the
>>3.1 branch with the fix by doing this:
>>
>>    git clone git://git.sip-router.org/sip-router -b 3.1
>>
>>Otherwise, here's the workaround I was using before:
>>
>>       while(mq_fetch("myq")) {
>>          ...
>>
>>          if(!defined $mqk(myq))
>>             break;
>>
>>           ...
>>        }
>>
>>Cheers,
>>
>>-- Alex
>
>Excellent, thanks for the quick response!
>
>Regards,
>- Brad




From gminghon at gmail.com  Wed Jun 29 03:30:47 2011
From: gminghon at gmail.com (MingHon)
Date: Wed, 29 Jun 2011 09:30:47 +0800
Subject: [SR-Users] two uac behind same nat one uac remain Cflag:: 0
Message-ID: <BANLkTikMOZqHh390FZb_E3BVFmvubY+vnw@mail.gmail.com>

Hi List,

both uac register successfully but the issue is the first uac who register
the Cflag remain 0.

im using nat_uac_test("19").

thank you in advance.

here is the ul show:

AOR:: 102
                Contact:: sip:102 at 175.136.221.60:5062 Q=
                        Expires:: 103
                        Callid:: 678936531 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: T22 7.3.0.50
                        Received:: sip:175.136.221.60:1056
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 192
                        Socket:: udp:60.53.82.184:5060
                        Methods:: 16383
        AOR:: 103
                Contact:: sip:103 at 175.136.221.60:5062 Q=
                        Expires:: 62
                        Callid:: 1621958925 at 175.136.221.60
                        Cseq:: 3
                        User-agent:: Yealink SIP-T18 18.0.0.70
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 0
                        Socket:: udp:60.53.82.184:5060
                        Methods:: 16383


and here is the wireshark on kamailio for ua 103 and 102.


E@< \<<5Rw REGISTER sip:aextddns.dyndns.info SIP/2.0
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766
From: "103" <sip:103 at 175.136.221.60>;tag=681558690
To: "103" <sip:103 at aextddns.dyndns.info>
Call-ID: 1621958925 at 175.136.221.60
CSeq: 2 REGISTER
Contact: <sip:103 at 175.136.221.60:5062>
Authorization: Digest username="103", realm="aextddns.dyndns.info",
nonce="Tgp7mU4Kem3Qqyd66hwhWgmU8bNpL463", uri="si$
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T18 18.0.0.70
Expires: 3600
Content-Length: 0

E@<5R<PSIP/2.0 200 OK
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766;rport=5062
From: "103" <sip:103 at 175.136.221.60>;tag=681558690
To: "103" <sip:103 at aextddns.dyndns.info
>;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.d24e
Call-ID: 1621958925 at 175.136.221.60
CSeq: 2 REGISTER
Contact: <sip:103 at 175.136.221.60:5062>;expires=120
Server: kamailio (3.1.4 (i386/linux))
Content-Length: 0

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

E@< j<<5R 7REGISTER sip:aextddns.dyndns.info SIP/2.0
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976
From: "102" <sip:102 at 175.136.221.60>;tag=544575245
To: "102" <sip:102 at aextddns.dyndns.info>
Call-ID: 678936531 at 175.136.221.60
CSeq: 2 REGISTER
Contact: <sip:102 at 175.136.221.60:5062>
Authorization: Digest username="102", realm="aextddns.dyndns.info",
nonce="Tgp8AU4KetWGb79MEU5MUHq+xA4xy+OW", uri="si$
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: T22 7.3.0.50
Expires: 3600
Content-Length: 0

E@<5R< {SIP/2.0 200 OK
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976;rport=1056
From: "102" <sip:102 at 175.136.221.60>;tag=544575245
To: "102" <sip:102 at aextddns.dyndns.info
>;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.5cb8
Call-ID: 678936531 at 175.136.221.60
CSeq: 2 REGISTER
Contact: <sip:102 at 175.136.221.60:5062>;expires=120;received="sip:
175.136.221.60:1056"
Server: kamailio (3.1.4 (i386/linux))
Content-Length: 0


-- 
Regards,

MingHon
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From jh at tutpro.com  Wed Jun 29 06:28:36 2011
From: jh at tutpro.com (Juha Heinanen)
Date: Wed, 29 Jun 2011 07:28:36 +0300
Subject: [SR-Users] Questions about LCR module
In-Reply-To: <SNT110-W11CE1518BA4DE438B67B63CD560@phx.gbl>
References: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>
	<SNT110-W61D8ABC08C48B487A15EFACD560@phx.gbl>
	<SNT110-W571E3FE9842B1B21139700CD560@phx.gbl>
	<SNT110-W11CE1518BA4DE438B67B63CD560@phx.gbl>
Message-ID: <19978.43508.688170.213114@sip.test.fi>

Fabian Borot writes:

> Then I tried the sercmd commands to reload and dump and I do not see
> the contents o the screen. Is this how is supposed to work now?. How
> do I know that the new values are being loaded into memory w/o
> restarting kamailio?

lcr.reload load the tables to memory and does not print
anything. lcr.dump_[gws|rules] prints what is in memory.

perhaps you didn't set lcr_count.  please read whole README carefully.

-- juha


From alhakeem at gmail.com  Wed Jun 29 14:06:12 2011
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Wed, 29 Jun 2011 13:06:12 +0100
Subject: [SR-Users] Proxy setup in a non-recursive mode
In-Reply-To: <4E0986B0.5010304@pernau.at>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEEwH9YGNtFIts3Q9/Tg/3YBAAAAAA==@gmail.com>
	<4E0986B0.5010304@pernau.at>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU+ivQ8AEAAAAEgu1O3wSeVGoyISV7u5rkEBAAAAAA==@gmail.com>

Thanks for the response.
Is Kamailio able to do a route lookup via HTTP get/post & pass the variables
received to it's routing table before routing the call ?


-----Original Message-----
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Klaus Darilion
Sent: Tuesday, June 28, 2011 8:46 AM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing
List
Subject: Re: [SR-Users] Proxy setup in a non-recursive mode

Being the "R-Proxy" is quite easy with Kamailio.

# lookup route and set new RURI
...
prefix("23430");

# add any custom headers to response
append_to_reply(.....);


# send 302 (new RURI will automatically added as Contact header)
sl_send_reply("302","Moved");
exit;


Of course Kamailio has no influence how the Gateway will behave on
reception of the 302 response.

regards
Klaus


Am 27.06.2011 19:03, schrieb Abdul Hakeem:
>
>
> Hello,
>
> Could anyone help with suggestions on how to setup a Proxy ( R-Proxy)
> that only does a route lookup and pass the parameter back to origination
> Proxy/Gateway A to forward the call based on the new parameter.
>
>
>
> On receipt of an invite from GW/Proxy A,   R- Proxy will do a route
> lookup for the called number 447988870290,
>
> If it finds a route then it should append the network prefix to the new
> INVITE or rewrite the INVITE 23430442088870290 in the 302 Moved
> Temporarily message sent back to GW/Proxy A in the Contact field and
> there are two CC-Diversion header fields included; at GW/Proxy (IP
> address) who originated the call and already has a route or dial plan
> for the new prefix and also another one for same at GW/Proxy (IP address).
>
>
>
> Is there any way to do this without creating a loop in the network ?
>
>
>
> Previously this setup can be achieved with the originating network
> having 2 GW/Proxies  i.e. GW/Proxy A  and GW/Proxy B. In the 302
> Temporarily moved message  the Contact will show GW/Proxy B, while the 2
> CC-Diversion header fields will add GW/Proxy B  and GW Proxy A.
>
>
>
> Regards,
>
> Abdul Hakeem
>
>
>
> _______________________________________________
> 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 spinov_evgeniy at intalisan.com  Wed Jun 29 14:42:22 2011
From: spinov_evgeniy at intalisan.com (Spinov Evgeniy)
Date: Wed, 29 Jun 2011 16:42:22 +0400
Subject: [SR-Users] MWI handling
Message-ID: <1309351342.2690.4.camel@tpadtest.timegroupme.local>

Hello.

Trying to understand how to work with the presence_mwi module. It seems
to me, that it's just for my purpose - inform subscribed peers about
waiting messages. But it do not export any functions, noting.

The scheme is simple: Peer->Kamailio->Asterisk.

Asterisk can accept voicemail and when the message is saved I want to
receive NOTIFY for subscribed peer. As I understand, peer should
register on Kamailio, kamailio should accept this subscription and send
subscription to asterisk. After this, need to proxy all NOTIFY to the
peer.

Anyway, cannot find any way to do that. How it's usually done, please,
advice.

Thanks in advance.



From vaad.fabi at gmail.com  Wed Jun 29 14:59:39 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Wed, 29 Jun 2011 15:59:39 +0300
Subject: [SR-Users] dispatcher gw state autochange
In-Reply-To: <C778BE90B341194A960B1EF37812EA6A0C3C9888@nasa-dtw-mbx02.nasa.cpwr.corp>
References: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>	<4E0A4A94.8090706@evaristesys.com>
	<C778BE90B341194A960B1EF37812EA6A0C3C9888@nasa-dtw-mbx02.nasa.cpwr.corp>
Message-ID: <4E0B21BB.1000306@gmail.com>

  Hi all,


Is it possible to autochange gateway's state from pending to active 
after gw get up ?

http://www.kamailio.org/docs/modules/1.4.x/dispatcher.html says only 
about ping caused autochange state from active to  pending, but nothing 
about change gw's state back.

-- 
Vadim F.



From gminghon at gmail.com  Wed Jun 29 15:56:50 2011
From: gminghon at gmail.com (MingHon)
Date: Wed, 29 Jun 2011 21:56:50 +0800
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
Message-ID: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>

will kamailio as proxy behind nat and UACs behind another nat work?

port forward sip and rtp done in the router.

UACs register successfully but no audio.

advertised_address = public_ip
advertised_port = sip

both define after the line of listen=public_ip

please advice.

thanks.

-- 
Regards,

MingHon
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From andrew.pogrebennyk at portaone.com  Wed Jun 29 16:04:35 2011
From: andrew.pogrebennyk at portaone.com (Andrew Pogrebennyk)
Date: Wed, 29 Jun 2011 17:04:35 +0300
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
In-Reply-To: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
References: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
Message-ID: <4E0B30F3.90501@portaone.com>

What are you using - rtpproxy, mediaproxy etc? There is an experimental 
patch for advertised_address support in rtpproxy on the internet, but 
I've never had a chance to try it out. You may want to try spce-v2.2, 
which comes with mediaproxy-ng supporting the advertised_address out of 
the box: http://www.sipwise.com/news/announcements/spce-v2_2-release/

On 29.06.2011 16:56, MingHon wrote:
> will kamailio as proxy behind nat and UACs behind another nat work?
>
> port forward sip and rtp done in the router.
>
> UACs register successfully but no audio.
>
> advertised_address = public_ip
> advertised_port = sip
>
> both define after the line of listen=public_ip
>
> please advice.
>
> thanks.

-- 
Sincerely,
Andrew Pogrebennyk


From fborot at hotmail.com  Wed Jun 29 16:16:14 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Wed, 29 Jun 2011 10:16:14 -0400
Subject: [SR-Users] Questions about LCR module
In-Reply-To: <SNT110-W11CE1518BA4DE438B67B63CD560@phx.gbl>
References: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>,
	<SNT110-W61D8ABC08C48B487A15EFACD560@phx.gbl>,
	<SNT110-W571E3FE9842B1B21139700CD560@phx.gbl>,
	<SNT110-W11CE1518BA4DE438B67B63CD560@phx.gbl>
Message-ID: <SNT110-W481CBD8E7544DBCBCE7071CD590@phx.gbl>



lcr.reload load the tables to memory and does not print
anything. lcr.dump_[gws|rules] prints what is in memory.

perhaps you didn't set lcr_count.  please read whole README carefully.

-- juha

you are right, I guess I did not have the module well configured

 

From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Subject: RE: Questions about LCR module
Date: Tue, 28 Jun 2011 16:48:49 -0400






























Thanks Juha
I populated the lcr tables by hand. This is how they look now, I have 2 lcr_ids [1 and 2], each instance has its own gw. And each instance has its own prefix to match for [4444 and 5555].
The idea is that a call comes in with R-URI 444413051234567, it should match the lcr_id= 200, then it should go out using gw 192.168.169.200 etc

mysql> select * from lcr_gw;
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
| id | lcr_id | gw_name    | ip_addr         | hostname          | port | params | uri_scheme | transport | strip | tag  | flags | defunct |
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
|  1 |    200 | switchvox  | 192.168.169.200 | pbx1.mydomain.com | 5060 | none   |          1 |         1 |     0 |      |     0 |       0 |
|  2 |    201 | switchvox2 | 192.168.169.201 | pbx1.mydomain.net | 5060 | none   |          1 |         1 |     0 |      |     0 |       0 |
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
mysql> select * from lcr_rule;
+----+--------+--------+----------+---------+---------+
| id | lcr_id | prefix | from_uri | stopper | enabled |
+----+--------+--------+----------+---------+---------+
|  1 |    200 | 4444   |          |       0 |       1 |
|  2 |    201 | 5555   |          |       0 |       1 |
+----+--------+--------+----------+---------+---------+
mysql> select * from lcr_rule_target;
+----+--------+---------+-------+----------+--------+
| id | lcr_id | rule_id | gw_id | priority | weight |
+----+--------+---------+-------+----------+--------+
|  1 |    200 |       1 |     1 |        5 |      1 |
|  2 |    201 |       2 |     2 |        5 |      1 |
+----+--------+---------+-------+----------+--------+

Then I tried the sercmd commands to reload and dump and I do not see the contents o the screen. Is this how is supposed to work now?. How do I know that the new values are being loaded into memory w/o restarting kamailio?

I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see
> huge differences on the LCR module.

there is a script somewhere (read wiki) that upgrades the tables.

> I have these main questions:
> 
> 1- does lcr_id replaces the grp_id?

not really.  lcr_ids denote independent lcr instances.

> 2- what happened to the fifo commands to add gws etc:

fifo commands to manipulate lcr tables are not supported by me.

-- juha



 

From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Subject: Questions about LCR module
Date: Mon, 27 Jun 2011 17:23:05 -0400









I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module.
I have these main questions:

1- does lcr_id replaces the grp_id?
2- what happened to the fifo commands to add gws etc:

 lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip>
           ............... add a gateway with flags, tag and strip ............
           ................(flags, tag, and strip are optional arguments) .....
 lcr rmgw  <gw_name> ..........................................................
           ............... delete a gateway
 lcr addroute <prefix> <from> <grp_id> <prio> .................................
           .............. add a route ( use '' to match anything in <from> )
 lcr rmroute  <prefix> <from> <grp_id> <prio> .................................
           .............. delete a route


3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload"  only the lcr tables

What would be the best way to use this module on this version?

Any help/hint would be greatly appreciated.
thank you

fborot





  		 	   		   		 	   		   		 	   		   		 	   		   		 	   		  
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From osas at voipembedded.com  Wed Jun 29 16:31:26 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 29 Jun 2011 10:31:26 -0400
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
In-Reply-To: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
References: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
Message-ID: <BANLkTimwQxOPaV=0QU6KhH1Lk=Zu5r2xjQ@mail.gmail.com>

It is possible to use kamailio and rtpproxy behind a NAT.
You need to properly craft the config to deal with the fact that
kamailio is behind NAT (proper Via, Route, Record-Route headers) and
you need to do port forwarding for SIP signalling and rtp.


Regards,
Ovidiu Sas

On Wed, Jun 29, 2011 at 9:56 AM, MingHon <gminghon at gmail.com> wrote:
> will kamailio as proxy behind nat and UACs behind another nat work?
> port forward sip and rtp done in the router.
> UACs register successfully but no audio.
> advertised_address = public_ip
> advertised_port = sip
> both define after the line of listen=public_ip
> please advice.
> thanks.
> --
> Regards,
>
> MingHon
>
> _______________________________________________
> 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 osas at voipembedded.com  Wed Jun 29 17:26:13 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 29 Jun 2011 11:26:13 -0400
Subject: [SR-Users] MWI handling
In-Reply-To: <1309351342.2690.4.camel@tpadtest.timegroupme.local>
References: <1309351342.2690.4.camel@tpadtest.timegroupme.local>
Message-ID: <BANLkTikfw4rJu292RazoL2ZqHki5FZPHMQ@mail.gmail.com>

If you are using asterisk for voicemail, you should let asterisk
notify the subscribers about mwi.
If you want to use kamailio for notifications, then you need to
configure asterisk to send PUBLISH requests to kamailio and kamailio
will notify the subscribers.


Regards,
Ovidiu Sas

On Wed, Jun 29, 2011 at 8:42 AM, Spinov Evgeniy
<spinov_evgeniy at intalisan.com> wrote:
> Hello.
>
> Trying to understand how to work with the presence_mwi module. It seems
> to me, that it's just for my purpose - inform subscribed peers about
> waiting messages. But it do not export any functions, noting.
>
> The scheme is simple: Peer->Kamailio->Asterisk.
>
> Asterisk can accept voicemail and when the message is saved I want to
> receive NOTIFY for subscribed peer. As I understand, peer should
> register on Kamailio, kamailio should accept this subscription and send
> subscription to asterisk. After this, need to proxy all NOTIFY to the
> peer.
>
> Anyway, cannot find any way to do that. How it's usually done, please,
> advice.
>
> 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 gminghon at gmail.com  Wed Jun 29 18:14:03 2011
From: gminghon at gmail.com (MingHon)
Date: Thu, 30 Jun 2011 00:14:03 +0800
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
In-Reply-To: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
References: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
Message-ID: <BANLkTi=XcAAypE0rW9OTpB=WoQxTze0s4Q@mail.gmail.com>

Hi,

Im using kamailio 3.1.4 as proxy and rtpproxy 1.2.1 in a same server and
asterisk 1.6 is at another server both server in the same lan.

im new at kamailio and rtpproxy. do you have to source or example mind to
share? i try googling but i cant find any.

thank you very much.


-- 
Regards,

MingHon
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From osas at voipembedded.com  Wed Jun 29 18:21:26 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 29 Jun 2011 12:21:26 -0400
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
In-Reply-To: <BANLkTi=XcAAypE0rW9OTpB=WoQxTze0s4Q@mail.gmail.com>
References: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
	<BANLkTi=XcAAypE0rW9OTpB=WoQxTze0s4Q@mail.gmail.com>
Message-ID: <BANLkTimX8w9mET5wQEeow0y0w0QSatY9=A@mail.gmail.com>

I experimented with it a while ago in my lab and the setup is no longer active.
There's no out of the box solution for this, as you need to craft
proper Route, Record-Route and Via headers based on the direction of
the call.  You need to understand how loose routing works in order to
set correct headers in outgoing requests.


Regards,
Ovidiu Sas

On Wed, Jun 29, 2011 at 12:14 PM, MingHon <gminghon at gmail.com> wrote:
> Hi,
> Im using kamailio 3.1.4 as proxy and rtpproxy 1.2.1 in a same server and
> asterisk 1.6 is at another server both server in the same lan.
> im new at kamailio and rtpproxy. do you have to source or example mind to
> share? i try googling but i cant find any.
> thank you very much.
>
> --
> Regards,
>
> MingHon
>
> _______________________________________________
> 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 fborot at hotmail.com  Tue Jun 28 19:12:07 2011
From: fborot at hotmail.com (Fabian Borot)
Date: Tue, 28 Jun 2011 13:12:07 -0400
Subject: [SR-Users] Questions about LCR module
In-Reply-To: <SNT110-W61D8ABC08C48B487A15EFACD560@phx.gbl>
References: <SNT110-W7870A4B52945798B518EBCD570@phx.gbl>,
	<SNT110-W61D8ABC08C48B487A15EFACD560@phx.gbl>
Message-ID: <SNT110-W571E3FE9842B1B21139700CD560@phx.gbl>
















Thanks Juha
I populated the lcr tables by hand. This is how they look now, I have 2 lcr_ids [1 and 2], each instance has its own gw. And each instance has its own prefix to match for [4444 and 5555].
The idea is that a call comes in with R-URI 444413051234567, it should match the lcr_id= 200, then it should go out using gw 192.168.169.200 etc

mysql> select * from lcr_gw;
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
| id | lcr_id | gw_name    | ip_addr         | hostname          | port | params | uri_scheme | transport | strip | tag  | flags | defunct |
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
|  1 |    200 | switchvox  | 192.168.169.200 | pbx1.mydomain.com | 5060 | none   |          1 |         1 |     0 |      |     0 |       0 |
|  2 |    201 | switchvox2 | 192.168.169.201 | pbx1.mydomain.net | 5060 | none   |          1 |         1 |     0 |      |     0 |       0 |
+----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+
mysql> select * from lcr_rule;
+----+--------+--------+----------+---------+---------+
| id | lcr_id | prefix | from_uri | stopper | enabled |
+----+--------+--------+----------+---------+---------+
|  1 |    200 | 4444   |          |       0 |       1 |
|  2 |    201 | 5555   |          |       0 |       1 |
+----+--------+--------+----------+---------+---------+
mysql> select * from lcr_rule_target;
+----+--------+---------+-------+----------+--------+
| id | lcr_id | rule_id | gw_id | priority | weight |
+----+--------+---------+-------+----------+--------+
|  1 |    200 |       1 |     1 |        5 |      1 |
|  2 |    201 |       2 |     2 |        5 |      1 |
+----+--------+---------+-------+----------+--------+

Then I tried the sercmd commands to reload and dump and I do not see the contents o the screen. Is this how is supposed to work now?.

with debug level set to 3 I do this "/usr/local/kamailio-3.1/sbin/sercmd lcr.dump_rules"
 and see these lines on the syslog:

Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1
Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl
Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl
Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called

if I do this: "/usr/local/kamailio-3.1/sbin/sercmd  lcr.reload"
I see these:


Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db.c:294]: connection 0x951b78 found in pool
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 5 columns returned from the query
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 40 bytes for result names at 0x9527b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 20 bytes for result types at 0x952610
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x9528b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[0]=[id]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952928
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[1]=[prefix]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951a98
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[2]=[from_uri]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x951b08
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[3]=[stopper]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952998
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[4]=[enabled]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:985]: loading, cycle 0 with <0> rows
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 5 columns
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x9528b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952928
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951a98
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x951b08
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952998
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x9527b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x952610
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 12 columns returned from the query
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 96 bytes for result names at 0x951ef8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 48 bytes for result types at 0x951c38
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x951b08
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[0]=[gw_name]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x951a98
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[1]=[ip_addr]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x952928
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[2]=[port]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[uri_scheme]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952610
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952610)[4]=[transport]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[5] at 0x952998
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[5]=[params]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[6] at 0x9527b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9527b8)[6]=[hostname]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[7] at 0x951fb8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[7]=[strip]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[8] at 0x952028
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[8]=[tag]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[9] at 0x952098
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[9]=[flags]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[10] at 0x952108
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952108)[10]=[defunct]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[11] at 0x951288
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951288)[11]=[id]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:169]: no rows returned from the query
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 12 columns
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x951b08
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x951a98
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x952928
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952610
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[5] at 0x952998
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[6] at 0x9527b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[7] at 0x951fb8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[8] at 0x952028
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[9] at 0x952098
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[10] at 0x952108
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[11] at 0x951288
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x951ef8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951c38
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 4 columns returned from the query
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x952998
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 16 bytes for result types at 0x951288
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x952098
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[0]=[rule_id]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952028
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[1]=[gw_id]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951fb8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[2]=[priority]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[weight]
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:1344]: loading, cycle 0 with <0> rows
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 4 columns
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x952098
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952028
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951fb8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x952998
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951288
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_pool.c:97]: connection still kept in the pool
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl
Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called

I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see
> huge differences on the LCR module.

there is a script somewhere (read wiki) that upgrades the tables.

> I have these main questions:
> 
> 1- does lcr_id replaces the grp_id?

not really.  lcr_ids denote independent lcr instances.

> 2- what happened to the fifo commands to add gws etc:

fifo commands to manipulate lcr tables are not supported by me.

-- juha



 

From: fborot at hotmail.com
To: sr-users at lists.sip-router.org
Subject: Questions about LCR module
Date: Mon, 27 Jun 2011 17:23:05 -0400









I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module.
I have these main questions:

1- does lcr_id replaces the grp_id?
2- what happened to the fifo commands to add gws etc:

 lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip>
           ............... add a gateway with flags, tag and strip ............
           ................(flags, tag, and strip are optional arguments) .....
 lcr rmgw  <gw_name> ..........................................................
           ............... delete a gateway
 lcr addroute <prefix> <from> <grp_id> <prio> .................................
           .............. add a route ( use '' to match anything in <from> )
 lcr rmroute  <prefix> <from> <grp_id> <prio> .................................
           .............. delete a route


3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload"  only the lcr tables

What would be the best way to use this module on this version?

Any help/hint would be greatly appreciated.
thank you

fborot





  		 	   		   		 	   		   		 	   		  
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From nicolas.dupuis at ovh.net  Tue Jun 28 15:51:53 2011
From: nicolas.dupuis at ovh.net (Marvin59)
Date: Tue, 28 Jun 2011 06:51:53 -0700 (PDT)
Subject: [SR-Users] Presence and Presence_mwi
Message-ID: <31946526.post@talk.nabble.com>


Hi all,

I am trying to configure kamailio-3.1.4 in order to send me MWI Notify when
there is change in messages.
I have already configure it, in order to use a FreeRadius to authenticate my
users but I have no idea to use Presence and Presence_mwi module.

I need this behavior :
 - The phone send me a subscribe, then if it authorize i send it the Notify
with the messages-waiting.
 - I must write myself the notify request because an external script will
tell me if there are messages waiting.

I have looked here
(http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence.html#id2614273
and http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence_mwi.html).

If someone has further informations ....

Thanks for your help.

Best regards,
 Marvin59


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From nicolas.dupuis at ovh.net  Wed Jun 29 14:54:29 2011
From: nicolas.dupuis at ovh.net (Marvin59)
Date: Wed, 29 Jun 2011 05:54:29 -0700 (PDT)
Subject: [SR-Users] MWI Notify Request
Message-ID: <31954500.post@talk.nabble.com>


Hi all,

I would like to know how kamailio sends Notify request by using the
handle_subscribe of presence module.
My purpose is to modify the request to add information for "message waiting
indication" like : Messages-Waiting: yes, Voice-Message: 2/0 (0/0), ...

I need your help !

Thanks in advance,
Best regards
 Marvin
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From nicolas.dupuis at ovh.net  Wed Jun 29 14:55:29 2011
From: nicolas.dupuis at ovh.net (Marvin59)
Date: Wed, 29 Jun 2011 05:55:29 -0700 (PDT)
Subject: [SR-Users] Presence and Presence_mwi
Message-ID: <31946526.post@talk.nabble.com>


Hi all,

I am trying to configure kamailio-3.1.4 in order to send me MWI Notify when
there is change in messages.
I have already configure it, in order to use a FreeRadius to authenticate my
users but I have no idea to use Presence and Presence_mwi module.

I need this behavior :
 - The phone send me a subscribe, then if it authorize i send it the Notify
with the messages-waiting.
 - I must write myself the notify request because an external script will
tell me if there are messages waiting.

I have looked here
(http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence.html#id2614273
and http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence_mwi.html).

If someone has further informations ....

Thanks for your help.

Best regards,
 Marvin59


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From Ricardo.Keigo at br.bosch.com  Wed Jun 29 19:35:48 2011
From: Ricardo.Keigo at br.bosch.com (Andrade Ricardo (CI/AFU1))
Date: Wed, 29 Jun 2011 19:35:48 +0200
Subject: [SR-Users] "select count(*)" with avp_db_query always returns null
In-Reply-To: <mailman.7110.1309356982.2672.sr-users@lists.sip-router.org>
References: <mailman.7110.1309356982.2672.sr-users@lists.sip-router.org>
Message-ID: <268C7AED54EDF64F808B167324D2CBF91A0815E133@SI-MBX18.de.bosch.com>

Hi list,

Perhaps this fits better into a bug report, but I'd like to know if somebody out there experienced a similar issue.
I am executing a "select count" query using the avp_db_query function, but it is not storing the results in any avp. Other queries are working fine.
I use db_mysql connected with a mysql 5.1 server.

I have tested this with kamailio-3.1.0 and kamailio-3.1.3, both didn't work. In an old box (version 1.3.x), the same query was returning the correct value.

---------------------------------------------------
Here is the case which is not working:

Code:
avp_delete("$avp(s:count)");
$var(ret) = avp_db_query("SELECT count(*) FROM subscriber where username='foo'",
    "$avp(s:count)");
xlog("L_INFO", "var(ret)=$var(ret) avp(s:count)=$avp(s:count)");


Output:
INFO: <script>: var(ret)=1 avp(s:count)=<null>

(notice the return code 1, which means that the query was successfull and there should be some output value stored in the avp, but it is <null>)

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

And here is one case which works as expected:

$var(ret) = avp_db_query("SELECT id FROM subscriber where username='foo'",
    "$avp(s:count)");

Output:
INFO: <script>: var(ret)=1 avp(s:count)=11

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

Thanks in advance for any feedback.

Best regards,
Ricardo Andrade.


From osas at voipembedded.com  Wed Jun 29 22:53:47 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 29 Jun 2011 16:53:47 -0400
Subject: [SR-Users] MWI Notify Request
In-Reply-To: <31954500.post@talk.nabble.com>
References: <31954500.post@talk.nabble.com>
Message-ID: <BANLkTikPS+8c9Q0NLQEL1FrmzSWdv6F6ow@mail.gmail.com>

All you need to do is send PUBLISH requests to kamailio.
If there's a subscription for the MWI for the account that PUBISH is
intended for, a NOTIFY will be sent out.

More info here:
http://www.faqs.org/rfcs/rfc3265.html
http://www.faqs.org/rfcs/rfc3903.html


Regards,
Ovidiu Sas

On Wed, Jun 29, 2011 at 8:54 AM, Marvin59 <nicolas.dupuis at ovh.net> wrote:
>
> Hi all,
>
> I would like to know how kamailio sends Notify request by using the
> handle_subscribe of presence module.
> My purpose is to modify the request to add information for "message waiting
> indication" like : Messages-Waiting: yes, Voice-Message: 2/0 (0/0), ...
>
> I need your help !
>
> Thanks in advance,
> Best regards
> ?Marvin
> --
> View this message in context: http://old.nabble.com/MWI-Notify-Request-tp31954500p31954500.html
> Sent from the OpenSER Users Mailing List mailing list archive at Nabble.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 osas at voipembedded.com  Wed Jun 29 22:53:47 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 29 Jun 2011 16:53:47 -0400
Subject: [SR-Users] MWI Notify Request
In-Reply-To: <31954500.post@talk.nabble.com>
References: <31954500.post@talk.nabble.com>
Message-ID: <BANLkTikPS+8c9Q0NLQEL1FrmzSWdv6F6ow@mail.gmail.com>

All you need to do is send PUBLISH requests to kamailio.
If there's a subscription for the MWI for the account that PUBISH is
intended for, a NOTIFY will be sent out.

More info here:
http://www.faqs.org/rfcs/rfc3265.html
http://www.faqs.org/rfcs/rfc3903.html


Regards,
Ovidiu Sas

On Wed, Jun 29, 2011 at 8:54 AM, Marvin59 <nicolas.dupuis at ovh.net> wrote:
>
> Hi all,
>
> I would like to know how kamailio sends Notify request by using the
> handle_subscribe of presence module.
> My purpose is to modify the request to add information for "message waiting
> indication" like : Messages-Waiting: yes, Voice-Message: 2/0 (0/0), ...
>
> I need your help !
>
> Thanks in advance,
> Best regards
> ?Marvin
> --
> View this message in context: http://old.nabble.com/MWI-Notify-Request-tp31954500p31954500.html
> Sent from the OpenSER Users Mailing List mailing list archive at Nabble.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 klaus.mailinglists at pernau.at  Wed Jun 29 23:13:24 2011
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Wed, 29 Jun 2011 23:13:24 +0200
Subject: [SR-Users] Proxy setup in a non-recursive mode
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU+ivQ8AEAAAAEgu1O3wSeVGoyISV7u5rkEBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEEwH9YGNtFIts3Q9/Tg/3YBAAAAAA==@gmail.com>
	<4E0986B0.5010304@pernau.at>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU+ivQ8AEAAAAEgu1O3wSeVGoyISV7u5rkEBAAAAAA==@gmail.com>
Message-ID: <4E0B9574.9090800@pernau.at>

Abdul Hakeem wrote:
> Thanks for the response.
> Is Kamailio able to do a route lookup via HTTP get/post 

Yes. Either use xhttp module (simple GET http request) or use a 
perl/python/lua script and do the http lookup in the script (for more 
complex http requests)


 > & pass the variables
> received to it's routing table before routing the call ?

there are no "routing tables", but you can implement any logic you need. 
E.g. the http response may contain a new Request URI which replaces the 
original one.

regards
Klaus


> 
> 
> -----Original Message-----
> From: sr-users-bounces at lists.sip-router.org
> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Klaus Darilion
> Sent: Tuesday, June 28, 2011 8:46 AM
> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing
> List
> Subject: Re: [SR-Users] Proxy setup in a non-recursive mode
> 
> Being the "R-Proxy" is quite easy with Kamailio.
> 
> # lookup route and set new RURI
> ...
> prefix("23430");
> 
> # add any custom headers to response
> append_to_reply(.....);
> 
> 
> # send 302 (new RURI will automatically added as Contact header)
> sl_send_reply("302","Moved");
> exit;
> 
> 
> Of course Kamailio has no influence how the Gateway will behave on
> reception of the 302 response.
> 
> regards
> Klaus
> 
> 
> Am 27.06.2011 19:03, schrieb Abdul Hakeem:
>>
>> Hello,
>>
>> Could anyone help with suggestions on how to setup a Proxy ( R-Proxy)
>> that only does a route lookup and pass the parameter back to origination
>> Proxy/Gateway A to forward the call based on the new parameter.
>>
>>
>>
>> On receipt of an invite from GW/Proxy A,   R- Proxy will do a route
>> lookup for the called number 447988870290,
>>
>> If it finds a route then it should append the network prefix to the new
>> INVITE or rewrite the INVITE 23430442088870290 in the 302 Moved
>> Temporarily message sent back to GW/Proxy A in the Contact field and
>> there are two CC-Diversion header fields included; at GW/Proxy (IP
>> address) who originated the call and already has a route or dial plan
>> for the new prefix and also another one for same at GW/Proxy (IP address).
>>
>>
>>
>> Is there any way to do this without creating a loop in the network ?
>>
>>
>>
>> Previously this setup can be achieved with the originating network
>> having 2 GW/Proxies  i.e. GW/Proxy A  and GW/Proxy B. In the 302
>> Temporarily moved message  the Contact will show GW/Proxy B, while the 2
>> CC-Diversion header fields will add GW/Proxy B  and GW Proxy A.
>>
>>
>>
>> Regards,
>>
>> Abdul Hakeem
>>
>>
>>
>> _______________________________________________
>> 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 gminghon at gmail.com  Thu Jun 30 04:08:22 2011
From: gminghon at gmail.com (MingHon)
Date: Thu, 30 Jun 2011 10:08:22 +0800
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
In-Reply-To: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
References: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
Message-ID: <BANLkTimvBfABbqYuKfZg0LWTeuSPpKpVdg@mail.gmail.com>

Hi,

Can you give me more detail, could you guide me how to?


thanks you very much..

-- 
Regards,

MingHon
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From osas at voipembedded.com  Thu Jun 30 06:36:38 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Thu, 30 Jun 2011 00:36:38 -0400
Subject: [SR-Users] kamailio + rtpproxy behind nat possible?
In-Reply-To: <BANLkTimvBfABbqYuKfZg0LWTeuSPpKpVdg@mail.gmail.com>
References: <BANLkTiniLW1bqcPdJwyhnW7CM6YVO9vCzA@mail.gmail.com>
	<BANLkTimvBfABbqYuKfZg0LWTeuSPpKpVdg@mail.gmail.com>
Message-ID: <BANLkTim+0Zc5Yi6nOG4VcEML4kwo7WvUbQ@mail.gmail.com>

As I mentioned before, based on direction of the call you may or may
not need to force an advertised IP/port (see the advertised_address
core parameter and set_advertised_address() core function)
Also, you will need to force the same IP in the Route/Record-Route
headers (see the rr module).
And also you will need to put proper IPs in SDP (see the rtpproxy module).
Start with a simple config, make calls, check all the SIP requests,
find out what's not properly set and fix the config.


Regards,
Ovidiu Sas

On Wed, Jun 29, 2011 at 10:08 PM, MingHon <gminghon at gmail.com> wrote:
> Hi,
> Can you give me more detail, could you guide me how to?
>
> thanks you very much..
>
> --
> Regards,
>
> MingHon
>
> _______________________________________________
> 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 Jun 30 09:30:59 2011
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Thu, 30 Jun 2011 10:30:59 +0300 (EEST)
Subject: [SR-Users] Table location, methods field
Message-ID: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>

Can someone send me bitmasks for the Subj.
Or just the reference to include file with comments in a source code.
I can't find it in all set of availible document's since the SER router. :(

I just have to know which UA except Grandstreams an CSipsample use RFC3311 :)

Thanks in advance.

Andrew O. Zhukov





From henning.westerholt at 1und1.de  Thu Jun 30 11:15:32 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Thu, 30 Jun 2011 11:15:32 +0200
Subject: [SR-Users] Table location, methods field
In-Reply-To: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>
References: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>
Message-ID: <201106301115.32212.henning.westerholt@1und1.de>

On Thursday 30 June 2011, Andrew O. Zhukov wrote:
> Can someone send me bitmasks for the Subj.
> Or just the reference to include file with comments in a source code.
> I can't find it in all set of availible document's since the SER router. :(
> 
> I just have to know which UA except Grandstreams an CSipsample use RFC3311
> :)

Hi Andrew,

they are defined in the parser, as power of two integers:

http://sip-router.org/doxygen/sip-router/branch/master/msg__parser_8h.html#a82f3c7234c5469f18d5e9fac5700cf27

Best regards,

Henning


From gnugk at telegroup.com.ua  Thu Jun 30 11:25:44 2011
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Thu, 30 Jun 2011 12:25:44 +0300
Subject: [SR-Users] Table location, methods field
In-Reply-To: <201106301115.32212.henning.westerholt@1und1.de>
References: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>
	<201106301115.32212.henning.westerholt@1und1.de>
Message-ID: <4E0C4118.2030101@telegroup.com.ua>

On 06/30/2011 12:15 PM, Henning Westerholt wrote:
> On Thursday 30 June 2011, Andrew O. Zhukov wrote:
>> Can someone send me bitmasks for the Subj.
>> Or just the reference to include file with comments in a source code.
>> I can't find it in all set of availible document's since the SER router. :(
>>
>> I just have to know which UA except Grandstreams an CSipsample use RFC3311
>> :)
>
> Hi Andrew,
>
> they are defined in the parser, as power of two integers:
>
> http://sip-router.org/doxygen/sip-router/branch/master/msg__parser_8h.html#a82f3c7234c5469f18d5e9fac5700cf27
>
> Best regards,
>
> Henning
>
Thanks !


From gnugk at telegroup.com.ua  Thu Jun 30 12:39:05 2011
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Thu, 30 Jun 2011 13:39:05 +0300
Subject: [SR-Users] Table location, methods field
In-Reply-To: <201106301115.32212.henning.westerholt@1und1.de>
References: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>
	<201106301115.32212.henning.westerholt@1und1.de>
Message-ID: <4E0C5249.1010403@telegroup.com.ua>

What I'm doing wrong?

select contact, User_agent, methods from location
where (methods & 2048) > 0 and user_agent like 'snom%';

contact                                  user_agent     methods
sip:3039200 at 10.0.1.55:2048;line=yd9qcfm9 snom320/7.3.14 7999


REGISTER sip:ipshka.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.55:2048;branch=z9hG4bK-sm9svrjytxjs;rport
From: "Andrew O. Zhukov" <sip:3039200 at ipshka.com>;tag=ir1p3z5ie6
To: "Andrew O. Zhukov" <sip:3039200 at ipshka.com>
Call-ID: 3c362d4fe75b-gi8h4zya7bbx
CSeq: 1170 REGISTER
Max-Forwards: 70
Contact: 
<sip:3039200 at 10.0.1.55:2048;line=yd9qcfm9>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:4dd6e733-e1cf-4966-b724-35ecca091e52>";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom320/7.3.14
Supported: gruu
Allow-Events: dialog
X-Real-IP: 10.0.1.55
Expires: 3600
Content-Length: 0
-----------------------------------------

So 7999 it's 1111100111111 binary

INVITE, 1
ACK,    3
CANCEL, 2
BYE,    4
INFO,   5
*
SUBSCRIBE, 7
NOTIFY, 8
MESSAGE, 9
OPTIONS, 10
PRACK, 11
*
REFER,  13

Something wrong !!!


On 06/30/2011 12:15 PM, Henning Westerholt wrote:
> On Thursday 30 June 2011, Andrew O. Zhukov wrote:
>> Can someone send me bitmasks for the Subj.
>> Or just the reference to include file with comments in a source code.
>> I can't find it in all set of availible document's since the SER router. :(
>>
>> I just have to know which UA except Grandstreams an CSipsample use RFC3311
>> :)
>
> Hi Andrew,
>
> they are defined in the parser, as power of two integers:
>
> http://sip-router.org/doxygen/sip-router/branch/master/msg__parser_8h.html#a82f3c7234c5469f18d5e9fac5700cf27
>
> Best regards,
>
> Henning
>



From gnugk at telegroup.com.ua  Thu Jun 30 12:44:24 2011
From: gnugk at telegroup.com.ua (Andrew O. Zhukov)
Date: Thu, 30 Jun 2011 13:44:24 +0300
Subject: [SR-Users] Table location, methods field
In-Reply-To: <201106301115.32212.henning.westerholt@1und1.de>
References: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>
	<201106301115.32212.henning.westerholt@1und1.de>
Message-ID: <4E0C5388.2090201@telegroup.com.ua>

Sorry,

You change it ...

enum request_method {
         METHOD_UNDEF=0,           /* 0 - --- */
         METHOD_INVITE=1,          /* 1 - 2^0 */
         METHOD_CANCEL=2,          /* 2 - 2^1 */
         METHOD_ACK=4,             /* 3 - 2^2 */
         METHOD_BYE=8,             /* 4 - 2^3 */
         METHOD_INFO=16,           /* 5 - 2^4 */
         METHOD_OPTIONS=32,        /* 6 - 2^5 */
         METHOD_UPDATE=64,         /* 7 - 2^6 */
         METHOD_REGISTER=128,      /* 8 - 2^7 */
         METHOD_MESSAGE=256,       /* 9 - 2^8 */
         METHOD_SUBSCRIBE=512,     /* 10 - 2^9 */
         METHOD_NOTIFY=1024,       /* 11 - 2^10 */
         METHOD_PRACK=2048,        /* 12 - 2^11 */
         METHOD_REFER=4096,        /* 13 - 2^12 */
         METHOD_PUBLISH=8192,      /* 14 - 2^13 */
         METHOD_OTHER=16384        /* 15 - 2^14 */
};



On 06/30/2011 12:15 PM, Henning Westerholt wrote:
> On Thursday 30 June 2011, Andrew O. Zhukov wrote:
>> Can someone send me bitmasks for the Subj.
>> Or just the reference to include file with comments in a source code.
>> I can't find it in all set of availible document's since the SER router. :(
>>
>> I just have to know which UA except Grandstreams an CSipsample use RFC3311
>> :)
>
> Hi Andrew,
>
> they are defined in the parser, as power of two integers:
>
> http://sip-router.org/doxygen/sip-router/branch/master/msg__parser_8h.html#a82f3c7234c5469f18d5e9fac5700cf27
>
> Best regards,
>
> Henning
>



From henning.westerholt at 1und1.de  Thu Jun 30 14:36:42 2011
From: henning.westerholt at 1und1.de (Henning Westerholt)
Date: Thu, 30 Jun 2011 14:36:42 +0200
Subject: [SR-Users] Table location, methods field
In-Reply-To: <4E0C5388.2090201@telegroup.com.ua>
References: <60694.193.28.184.10.1309419059.squirrel@mail.telegroup.com.ua>
	<201106301115.32212.henning.westerholt@1und1.de>
	<4E0C5388.2090201@telegroup.com.ua>
Message-ID: <201106301436.42942.henning.westerholt@1und1.de>

On Thursday 30 June 2011, Andrew O. Zhukov wrote:
> Sorry,
> 
> You change it ...
> 
> enum request_method {
>          METHOD_UNDEF=0,           /* 0 - --- */
>          METHOD_INVITE=1,          /* 1 - 2^0 */
>          METHOD_CANCEL=2,          /* 2 - 2^1 */
>          METHOD_ACK=4,             /* 3 - 2^2 */
>          METHOD_BYE=8,             /* 4 - 2^3 */
>          METHOD_INFO=16,           /* 5 - 2^4 */
>          METHOD_OPTIONS=32,        /* 6 - 2^5 */
>          METHOD_UPDATE=64,         /* 7 - 2^6 */
>          METHOD_REGISTER=128,      /* 8 - 2^7 */
>          METHOD_MESSAGE=256,       /* 9 - 2^8 */
>          METHOD_SUBSCRIBE=512,     /* 10 - 2^9 */
>          METHOD_NOTIFY=1024,       /* 11 - 2^10 */
>          METHOD_PRACK=2048,        /* 12 - 2^11 */
>          METHOD_REFER=4096,        /* 13 - 2^12 */
>          METHOD_PUBLISH=8192,      /* 14 - 2^13 */
>          METHOD_OTHER=16384        /* 15 - 2^14 */
> };

Hi Andrew,

the docs are a bit misleading, i'll have a look and probably remove the 
initial number. The 2^n values are the correct ones.

Best regards,

Henning


From osas at voipembedded.com  Thu Jun 30 15:56:19 2011
From: osas at voipembedded.com (Ovidiu Sas)
Date: Thu, 30 Jun 2011 09:56:19 -0400
Subject: [SR-Users] MWI handling
In-Reply-To: <1309440636.1996.3.camel@tpadtest.timegroupme.local>
References: <1309440636.1996.3.camel@tpadtest.timegroupme.local>
Message-ID: <BANLkTin_Y6a2-hk9kq1Yy1KqFA-SoMUJJw@mail.gmail.com>

If you are using asterisk as a voicemail server, let asterisk to
handle the SUBSCRIBE and don't involve the kamailio presence server in
this process.
If you really want to use kamailio as a mwi presence server, then you
will need to build some external scripts for pooling the asterisk mwi
db and send PUBLISH out.

I would stick with asterisk handling the mwi subscription and
configure kamailio as a relay only for those events.


Regards,
Ovidiu Sas


On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy
<spinov_evgeniy at intalisan.com> wrote:
> This makes the main problem as in documents I've found on asterisk is
> that it doesn't support PUBLISH, i.e. there is no way to force sending
> them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme.
>
> Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio
> about MWI? If yes, how? Cause right now, for me, it looks like a theory,
> cause no mechanics to force asterisk send PUBLISHes.
>
> Thank you.
>
> Regards,
> Evgeniy Spinov
>
>
>> If you are using asterisk for voicemail, you should let asterisk
>> notify the subscribers about mwi.
>> If you want to use kamailio for notifications, then you need to
>> configure asterisk to send PUBLISH requests to kamailio and kamailio
>> will notify the subscribers.
>
>
>> Regards,
>> Ovidiu Sas
>
>


From spinov_evgeniy at intalisan.com  Thu Jun 30 15:41:05 2011
From: spinov_evgeniy at intalisan.com (Spinov Evgeniy)
Date: Thu, 30 Jun 2011 17:41:05 +0400
Subject: [SR-Users] MWI handling
In-Reply-To: <1309440636.1996.3.camel@tpadtest.timegroupme.local>
References: <1309440636.1996.3.camel@tpadtest.timegroupme.local>
Message-ID: <1309441265.1996.4.camel@tpadtest.timegroupme.local>

This makes the main problem as in documents I've found on asterisk is
that it doesn't support PUBLISH, i.e. there is no way to force sending
them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme.

Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio
about MWI? If yes, how? Cause right now, for me, it looks like a theory,
cause no mechanics to force asterisk send PUBLISHes.

Thank you.

Regards,
Evgeniy Spinov


> If you are using asterisk for voicemail, you should let asterisk
> notify the subscribers about mwi.
> If you want to use kamailio for notifications, then you need to
> configure asterisk to send PUBLISH requests to kamailio and kamailio
> will notify the subscribers.


> Regards,
> Ovidiu Sas



From spinov_evgeniy at intalisan.com  Thu Jun 30 15:48:39 2011
From: spinov_evgeniy at intalisan.com (Spinov Evgeniy)
Date: Thu, 30 Jun 2011 17:48:39 +0400
Subject: [SR-Users] MWI handling
In-Reply-To: <BANLkTikNdTsegMEYHDVUaBJTYCn8BZ_cxg@mail.gmail.com>
References: <1309440636.1996.3.camel@tpadtest.timegroupme.local>
	<BANLkTikNdTsegMEYHDVUaBJTYCn8BZ_cxg@mail.gmail.com>
Message-ID: <1309441719.1996.7.camel@tpadtest.timegroupme.local>

I've done that in this fashion: Kamailio is only proxing SUBSCRIBEs to
Asterisk and it's replying with NOTIFYs to Kamailio, Kamailio is just
proxying reply to UAC. So Kamailio is transparent for MWI.

But it looks like spike. I'd better send PUBLISHes from Asterisk to
Kamailio for all the clients and Kamailio reply with NOTIFY to
subscribed ones. This is like Kamailio MWI module was designed.

Anyway, thanks for the help. Scripts for making PUBLISHES is spike even
more.

On Thu, 2011-06-30 at 09:53 -0400, Ovidiu Sas wrote:
> If you are using asterisk as a voicemail server, let asterisk to
> handle the SUBSCRIBE and don't involve the kamailio presence server in
> this process.
> If you really want to use kamailio as a mwi presence server, then you
> will need to build some external scripts for pooling the asterisk mwi
> db and send PUBLISH out.
> 
> I would stick with asterisk handling the mwi subscription and
> configure kamailio as a relay only for those events.
> 
> 
> Regards,
> Ovidiu Sas
> 
> On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy
> <spinov_evgeniy at intalisan.com> wrote:
> > This makes the main problem as in documents I've found on asterisk is
> > that it doesn't support PUBLISH, i.e. there is no way to force sending
> > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme.
> >
> > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio
> > about MWI? If yes, how? Cause right now, for me, it looks like a theory,
> > cause no mechanics to force asterisk send PUBLISHes.
> >
> > Thank you.
> >
> > Regards,
> > Evgeniy Spinov
> >
> >
> >> If you are using asterisk for voicemail, you should let asterisk
> >> notify the subscribers about mwi.
> >> If you want to use kamailio for notifications, then you need to
> >> configure asterisk to send PUBLISH requests to kamailio and kamailio
> >> will notify the subscribers.
> >
> >
> >> Regards,
> >> Ovidiu Sas
> >
> >




From lau.testi at gmail.com  Thu Jun 30 16:10:30 2011
From: lau.testi at gmail.com (laura testi)
Date: Thu, 30 Jun 2011 16:10:30 +0200
Subject: [SR-Users]
	=?windows-1252?q?xcap_server_-_http_load=96balancer=3A?=
	=?windows-1252?q?_xcap_authentication_problem?=
In-Reply-To: <4E0A20B8.1090203@gmail.com>
References: <BANLkTim2D87vqzFy59OQ7iTeORMDrosAhg@mail.gmail.com>
	<4E0A20B8.1090203@gmail.com>
Message-ID: <BANLkTimxb2bR=G5QXDo0n1dP0rJTvS0N6g@mail.gmail.com>

Thank you very much Daniel! It works!

Following your suggestion, I have set the following parameters in the
kamailio.cfg in both servers:

modparam("auth", "secret", "samesecretparam")
modparam("auth", "nonce_expire", 1800)
modparam("auth", "nonce_auth_max_drift", 300)
modparam("auth", "one_time_nonce", 0)


Now all xcap requests are authenticated successfully and works! Thanks
again!

After some tests, now we are facing another problem with multiple presence
servers integrated with XCAP servers in integrated mode. We have configured
2 kamailio servers as both presence server and xcap server with the same
backend DB in another machine (thank you for your great guide on
http://kb.asipto.com/kamailio:presence:k31-made-simple). In front we have
setup a dispatcher for SIP load balancing and apache for http (xcap) load
balancing.

It seems the presence server handle the publish/subscribe/notify events with
the following tables:
- active_watchers (using local cache and update the local cache to db
periodically)
- presentity (using local cache and update to db immediately)
- watchers (db)
- xcap (db)

we have meet 2 problems:

   1. when a subscription is created in one server (in active_watchers in
   local cache) and the publish related to the same subscription goes to
   another server (presentity), with fallback to db, only the first publish
   works (it generates a notify with C-SEQ increased by one). When it receives
   another publish, it always generate the notify with the same C-SEQ, and the
   SIP Client ignore it. The problem is that the subscription is in the cache
   of another server and it periodically update it to db. When the publish
   related to this subscription goes to another server, the C-SEQ in the local
   cache is not increased. To workaround it, we have configure the Dispatcher
   to use hashing over URI for all presence related messages. In this way, it
   seems all messages related to the same subscription go to the same presence
   server, so the c-seq is increased correctly. Is it correct way to work with
   multiple presence server?
   2. pres_refresh_watchers triggered by xcap message in another presence
   server: with the hashing over to uri in dispacther workaround, it seems
   solve the first problem for SIP/SIMPLE messages, but we have the same kind
   of problem for xcap message. For example, a subscription is in a local cache
   of one server, and the incomming xcap message related to the same
   subscription goes to another server, and this message trigger the
   pres_update_watchers pres_refresh_watchers presence functions from the
   configuration script in the server where there is no subscription in the
   local cache, then it send the wrong notify message. This can happen when a
   user add/remove a contact, and the SUBSCRIBE goes to one server and XCAP PUT
   goes to another server. Unfortunately there is no DB mode only in PRESENCE
   module like REGISTRAR module. The fallback to db can't help either for point
   1 or for point 2. Can you help please?



Thank you in advanced!



Best Regards,
Laura


On Tue, Jun 28, 2011 at 8:43 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
>
>
> On 6/28/11 6:33 PM, laura testi wrote:
>
> Hello all,
>
> I?m using two Kamailio Presence Servers that are also XCAP Servers for the
> presence and resources management and the Kamailio Dispatcher in order to
> balance the SIP requestes coming from SIP Client.
>
> Then I?m trying to configure a HTTP  load?balancer for the xcap/http
> traffic.
>
> I?m testing both Apache and INginx HTTP proxy but I?m facing with a problem
> related with the Xcap authentication:
>
> when a client sends the first xcap request to the http load-balancer, it
> forwards the request to one of the configured Xcap Server that replies with
> the 401 response in order to let the client authenticates. So, the client
> sends the second request with the Digest authentication to the http
> load-balancer. But, sometimes, the http load-balancer forwards the
> authenticate request to the second Xcap Server that is not able to
> authenticate the request.  Then the client receive the ?Unauthorized?
> response?
>
>
>
> Could you please give us some hints about this problem?
>
>
> you have to set the same value to secret parameter of auth module:
> http://kamailio.org/docs/modules/stable/modules/auth.html#auth.secret
>
> Also be sure the two servers have the same time (use ntp or so) and the
> one_time_nonce parameter to auth module is off (not enabled).
>
> Then it should work.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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From djover at umh.es  Thu Jun 30 19:43:57 2011
From: djover at umh.es (Dominguez Jover, Ricardo)
Date: Thu, 30 Jun 2011 19:43:57 +0200
Subject: [SR-Users] Internet RTP filters
Message-ID: <46E06B3F28AD344DBE53103DE027DC9D059FA5FB@exmailer2.umhnet.es>

Hi everybody,

First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us.

We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine.

This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up).

In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams:

- RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port)
- RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port)

As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine.

However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them.

Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result.

How could we solve this?

Thanks in advance and best regards,

Ricardo Dominguez
Universidad Miguel Hern?ndez de Elche





From andreaskaschner at mac.com  Thu Jun 30 22:04:21 2011
From: andreaskaschner at mac.com (andreas kaschner)
Date: Thu, 30 Jun 2011 22:04:21 +0200
Subject: [SR-Users] Internet RTP filters
In-Reply-To: <46E06B3F28AD344DBE53103DE027DC9D059FA5FB@exmailer2.umhnet.es>
References: <46E06B3F28AD344DBE53103DE027DC9D059FA5FB@exmailer2.umhnet.es>
Message-ID: <4530A201-A70E-42A0-8259-3989A686FBC8@mac.com>

hi, have you tried TLS with Minimum encryption?
regards Andreas



On 30. juni 2011, at 19:43, "Dominguez Jover, Ricardo" <djover at umh.es> wrote:

> Hi everybody,
> 
> First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us.
> 
> We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine.
> 
> This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up).
> 
> In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams:
> 
> - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port)
> - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port)
> 
> As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine.
> 
> However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them.
> 
> Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result.
> 
> How could we solve this?
> 
> Thanks in advance and best regards,
> 
> Ricardo Dominguez
> Universidad Miguel Hern?ndez de Elche
> 
> 
> 
> 
> _______________________________________________
> 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 djover at umh.es  Thu Jun 30 23:35:30 2011
From: djover at umh.es (Dominguez Jover, Ricardo)
Date: Thu, 30 Jun 2011 23:35:30 +0200
Subject: [SR-Users] Internet RTP filters
References: <46E06B3F28AD344DBE53103DE027DC9D059FA5FB@exmailer2.umhnet.es>
	<4530A201-A70E-42A0-8259-3989A686FBC8@mac.com>
Message-ID: <46E06B3F28AD344DBE53103DE027DC9D01FBE7AB@exmailer2.umhnet.es>

Hi Andreas, I've tried with TLS (using TCP port 5061) with no RTP encryption and with SRTP, but I don't know what minimum encryption means.

Regards,
Ricardo


-----Mensaje original-----
De: sr-users-bounces at lists.sip-router.org en nombre de andreas kaschner
Enviado el: jue 30/06/2011 22:04
Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -	UsersMailingList
Asunto: Re: [SR-Users] Internet RTP filters
 
hi, have you tried TLS with Minimum encryption?
regards Andreas



On 30. juni 2011, at 19:43, "Dominguez Jover, Ricardo" <djover at umh.es> wrote:

> Hi everybody,
> 
> First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us.
> 
> We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine.
> 
> This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up).
> 
> In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams:
> 
> - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port)
> - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port)
> 
> As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine.
> 
> However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them.
> 
> Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result.
> 
> How could we solve this?
> 
> Thanks in advance and best regards,
> 
> Ricardo Dominguez
> Universidad Miguel Hern?ndez de Elche
> 
> 
> 
> 
> _______________________________________________
> 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 carsten at ng-voice.com  Thu Jun 30 15:20:35 2011
From: carsten at ng-voice.com (Carsten Bock)
Date: Thu, 30 Jun 2011 15:20:35 +0200
Subject: [SR-Users] dispatcher gw state autochange
In-Reply-To: <4E0B21BB.1000306@gmail.com>
References: <C778BE90B341194A960B1EF37812EA6A0C3C9835@nasa-dtw-mbx02.nasa.cpwr.corp>
	<4E0A4A94.8090706@evaristesys.com>
	<C778BE90B341194A960B1EF37812EA6A0C3C9888@nasa-dtw-mbx02.nasa.cpwr.corp>
	<4E0B21BB.1000306@gmail.com>
Message-ID: <BANLkTi=R=QH8f2wkZVJX_yLW=O5ZgB1bHg@mail.gmail.com>

In the later versions, you could use the following parameter:

3.23. ds_probing_mode (int)

Controls what gateways are tested to see if they are reachable. If set
to 0, only the gateways with state PROBING are tested, if set to 1,
all gateways are tested. If set to 1 and the response is 408
(timeout), an active gateway is set to PROBING state.

Default value is ?0?.

Example 1.24. Set the ?ds_probing_mode? parameter

 ...
 modparam("dispatcher", "ds_probing_mode", 1)
 ...

In this case, the Dispatcher module would send a OPTIONS-request to
all the gateways and if it replies with a "200 OK", enable them
again....

Carsten

2011/6/29 vaad.fabi at gmail.com <vaad.fabi at gmail.com>:
> ?Hi all,
>
>
> Is it possible to autochange gateway's state from pending to active after gw
> get up ?
>
> http://www.kamailio.org/docs/modules/1.4.x/dispatcher.html says only about
> ping caused autochange state from active to ?pending, but nothing about
> change gw's state back.
>
> --
> Vadim F.
>
>
> _______________________________________________
> 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
http://www.ng-voice.com
mailto:carsten at ng-voice.com

Schomburgstr. 80
22767 Hamburg
Germany

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.6.1310038933.2669.sr-users@lists.sip-router.org>

when the prefixes are overlaping in terms of they are the same but with
different attrs like :
2       50   ^38.+              porvider_A
2       50   ^3802.+          porvider_B
*2       50   ^3806.+          porvider_C*
*2       50   ^3806.+          porvider_D*
*
*
here i understand that I should use priorities.

Does anyone knnows how this works ??

Thank you in advance
Alex
*
*

--0016363b9ce0c869ca04a77934a4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Dear all,<div><br></div><div>I have these 3 entries in the dialplan:</div><=
div><br></div><div>dpid =A0 pr =A0 match_exp =A0 =A0 =A0attr</div><div><br>=
</div><div>2 =A0 =A0 =A0 50 =A0 ^38.+ =A0 =A0 =A0 =A0 =A0 =A0 =A0porvider_A=
</div><div>2 =A0 =A0 =A0 50 =A0 ^3802.+ =A0 =A0 =A0 =A0 =A0porvider_B</div>
<div>2 =A0 =A0 =A0 50 =A0 ^3806.+ =A0 =A0 =A0 =A0 =A0porvider_C</div><div><=
br></div><div>with the fifo command :</div><div><br></div><div><div><b>kamc=
tl fifo dp_translate 2 3806328881</b></div><div>Output::=A0</div><div>ATTRI=
BUTES::<b>=A0porvider_A</b></div>
</div><div><br></div><div>I would expect that the dialplan module would cho=
ose the third choice which is the correct one for this call.</div><div><br>=
</div><div>From the dialplan documentation i understand that the priority i=
s needed when the prefixes are overlaping in terms of they are the same but=
 with different attrs like :</div>
<div><div>2 =A0 =A0 =A0 50 =A0 ^38.+ =A0 =A0 =A0 =A0 =A0 =A0 =A0porvider_A<=
/div><div>2 =A0 =A0 =A0 50 =A0 ^3802.+ =A0 =A0 =A0 =A0 =A0porvider_B</div><=
div><b>2 =A0 =A0 =A0 50 =A0 ^3806.+ =A0 =A0 =A0 =A0 =A0porvider_C</b></div>=
</div><div><b>2 =A0 =A0 =A0 50 =A0 ^3806.+ =A0 =A0 =A0 =A0 =A0porvider_D</b=
></div>
<div><b><br></b></div><div>here i understand that I should use priorities.<=
/div><div><br></div><div>Does anyone knnows how this works ??</div><div><br=
></div><div>Thank you in advance</div><div>Alex</div><div><b><br></b></div>

--0016363b9ce0c869ca04a77934a4--


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.7.1310397071.2669.sr-users@lists.sip-router.org>

> {s.strip,len}
> Return string after removing starting 'len' characters. Parameter 'len' can
> be positive integer or pseudo-variable holding a positive integer.


I tried to use pvar holding len but got an error.

Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: pv
[pv_trans.c:434]: select cannot get p1
Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: <core>
[lvalue.c:351]: non existing right pvar

This is part of my config file:
        $avp(varname) = '1';            <---- pseudo variable holding a
positive integer
        $avp(result) = $(fU{s.strip,$avp(varname)});            <----
s.strip

How can i  use variables with transformation ?

--20cf3054a5a5894f2a04a7cc973d
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>From the transformations documentation:</div><blockquote class=3D"gmai=
l_quote" style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; m=
argin-left: 0.8ex; border-left-width: 1px; border-left-color: rgb(204, 204,=
 204); border-left-style: solid; padding-left: 1ex; ">
{s.strip,len}<br>Return string after removing starting &#39;len&#39; charac=
ters. Parameter &#39;len&#39; can be positive integer or pseudo-variable ho=
lding a positive integer.</blockquote><div><br></div><div>I tried to use pv=
ar holding len but got an error.</div>
<div><br></div><div><div>Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905=
]: ERROR: pv [pv_trans.c:434]: select cannot get p1=A0</div><div>Jul 11 14:=
51:37 localhost /usr/sbin/kamailio[16905]: ERROR: &lt;core&gt; [lvalue.c:35=
1]: non existing right pvar=A0</div>
</div><div><br></div><div>This is part of my config file:</div><div>=A0 =A0=
 =A0 =A0 $avp(varname) =3D &#39;1&#39;; =A0 =A0 =A0 =A0 =A0 =A0&lt;---- pse=
udo variable holding a positive integer</div><div><div>=A0 =A0 =A0 =A0 $avp=
(result) =3D $(fU{s.strip,$avp(varname)}); =A0 =A0 =A0 =A0 =A0 =A0&lt;---- =
s.strip</div>
</div><div><br></div><div>How can i =A0use variables with transformation ?<=
/div>

--20cf3054a5a5894f2a04a7cc973d--


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.8.1310397283.2669.sr-users@lists.sip-router.org>

>>{s.strip,len}
>>Return string after removing starting 'len' characters. Parameter 'len' c=
an be positive integer or pseudo-variable holding a positive integer.

I tried to use pvar holding len but got an error:
Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: pv
[pv_trans.c:434]: select cannot get p1
Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: <core>
[lvalue.c:351]: non existing right pvar

This is part of my config file:
=A0 =A0 =A0 =A0 $avp(varname) =3D '1'; =A0 =A0 =A0 =A0 =A0 =A0<---- pseudo =
variable holding
a positive integer
=A0 =A0 =A0 =A0 $avp(result) =3D $(fU{s.strip,$avp(varname)}); =A0 =A0 =A0 =
=A0 =A0 =A0<---- s.strip

How can i =A0use variables with transformation ?


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.9.1310569032.2669.sr-users@lists.sip-router.org>

to the INVITE as it passes through the proxy- is this correct?

If so, can I ask how it handles situations where the UAS does not resend
these headers in subsequent requests?

Pete

--000e0cd4b55ccaa78104a7f4a0e6
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi<div><br></div><div>I have been researching the topoh module which I unde=
rstand is stateless and can persist across a server restart.</div><div><br>=
</div><div>From what I can gather it does this by adding some extra, encryp=
ted headers to the INVITE as it passes through the proxy- is this correct?<=
/div>
<div><br></div><div>If so, can I ask how it handles situations where the UA=
S does not resend these headers in subsequent requests?</div><div><br></div=
><div>Pete</div>

--000e0cd4b55ccaa78104a7f4a0e6--


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.10.1311261028.2669.sr-users@lists.sip-router.org>

mandatory hence it could be generated by a machine - whose
identity/location is not needed.

Regards,
Marius

------------------------------------------------------------------------
*From:* Daniel-Constantine Mierla <a class="moz-txt-link-rfc2396E" href="mailto:miconda at gmail.com">&lt;miconda at gmail.com&gt;</a>
*To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)
-Users Mailing List <a class="moz-txt-link-rfc2396E" href="mailto:sr-users at lists.sip-router.org">&lt;sr-users at lists.sip-router.org&gt;</a>
*Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
UsersMailing List <a class="moz-txt-link-rfc2396E" href="mailto:sr-users at lists.sip-router.org">&lt;sr-users at lists.sip-router.org&gt;</a>
*Sent:* Wednesday, July 20, 2011 6:53 PM
*Subject:* Re: [SR-Users] Kamailio presence server problem with
PostgreSQL

Hello,

Thanks for troubleshooting further. What intrigues me is that the
error message from syslog is printed from store result function,
which is used after doing a select query. So something else might
have to be reviewed as well.

Anyhow, missing sender value should be checked, since it is a
constraint for non-null. Interesting it was not risen with mysql.

Cheers,
Daniel

On Jul 20, 2011, at 5:46 PM, Roberto Fichera &lt;<a class="moz-txt-link-abbreviated" href="mailto:kernel at tekno-soft.it">kernel at tekno-soft.it</a>
<a class="moz-txt-link-rfc2396E" href="mailto:kernel at tekno-soft.it">&lt;mailto:kernel at tekno-soft.it&gt;</a>&gt; wrote:

</pre>
              <blockquote type="cite">
                <pre wrap="">On 07/20/2011 05:34 PM, Roberto Fichera wrote:
</pre>
                <blockquote type="cite">
                  <pre wrap="">On 07/20/2011 05:18 PM, Klaus Darilion wrote:
</pre>
                  <blockquote type="cite">
                    <pre wrap="">Am 20.07.2011 17:00, schrieb Roberto Fichera:
</pre>
                    <blockquote type="cite">
                      <pre wrap="">ERROR:  null value in column "sender" violates not-null constraint
STATEMENT:  insert into presentity
(domain,username,event,etag,expires,body,received_time ) values

</pre>
                    </blockquote>
                  </blockquote>
                </blockquote>
              </blockquote>
              <pre wrap="">('192.168.2.92','roberto','presence','a.1311173612.4970.2.0',1311177224,'&lt;?xml
</pre>
              <blockquote type="cite">
                <blockquote type="cite">
                  <blockquote type="cite">
                    <blockquote type="cite">
                      <pre wrap="">version="1.0" encoding="UTF-8" standalone="no"?&gt;&lt;presence
xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model"
xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
entity=<a class="moz-txt-link-rfc2396E" href="mailto:sip:roberto at 192.168.2.92">"sip:roberto at 192.168.2.92"</a>&gt;&lt;dm:person
id="p3619"&gt;&lt;rpid:activities/&gt;&lt;/dm:person&gt;&lt;tuple

</pre>
                    </blockquote>
                  </blockquote>
                </blockquote>
              </blockquote>
              <pre wrap="">id="t9285"&gt;&lt;status&gt;&lt;basic&gt;open&lt;/basic&gt;&lt;/status&gt;&lt;contact&gt;<a class="moz-txt-link-abbreviated" href="mailto:sip:roberto at 192.168.2.92">sip:roberto at 192.168.2.92</a>&lt;/contact&gt;&lt;note&gt;Online&lt;/note&gt;&lt;/tuple&gt;&lt;/presence&gt;',1311173624)
</pre>
              <blockquote type="cite">
                <blockquote type="cite">
                  <blockquote type="cite">
                    <pre wrap="">Seems like the statement does not insert the "sender" although it is
mandatory. If "sender" is really mandatory, then it seems the
</pre>
                  </blockquote>
                </blockquote>
              </blockquote>
              <pre wrap="">statement
</pre>
              <blockquote type="cite">
                <blockquote type="cite">
                  <blockquote type="cite">
                    <pre wrap="">is buggy.
</pre>
                  </blockquote>
                  <pre wrap="">how does look the corresponding presentity mysql table, does it
</pre>
                </blockquote>
              </blockquote>
              <pre wrap="">has the same
</pre>
              <blockquote type="cite">
                <blockquote type="cite">
                  <pre wrap="">constraint or not?
</pre>
                </blockquote>
                <pre wrap="">Anyway I can confirm that removing the constraint the problem
of presence backed by postgresql seems solved.

</pre>
                <blockquote type="cite">
                  <blockquote type="cite">
                    <pre wrap="">regards
Klaus

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
</pre>
                  </blockquote>
                </blockquote>
              </blockquote>
              <pre wrap="">mailing list
</pre>
              <blockquote type="cite">
                <blockquote type="cite">
                  <blockquote type="cite">
                    <pre wrap=""><a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:sr-users at lists.sip-router.org">&lt;mailto:sr-users at lists.sip-router.org&gt;</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>

</pre>
                  </blockquote>
                  <pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
</pre>
                </blockquote>
              </blockquote>
              <pre wrap="">list
</pre>
              <blockquote type="cite">
                <blockquote type="cite">
                  <pre wrap=""><a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:sr-users at lists.sip-router.org">&lt;mailto:sr-users at lists.sip-router.org&gt;</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>

</pre>
                </blockquote>
                <pre wrap="">
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:sr-users at lists.sip-router.org">&lt;mailto:sr-users at lists.sip-router.org&gt;</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
              </blockquote>
              <pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:sr-users at lists.sip-router.org">&lt;mailto:sr-users at lists.sip-router.org&gt;</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>




_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
            </blockquote>
            <pre wrap="">-- 
Daniel-Constantin Mierla -- <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
Kamailio Advanced Training, Oct 10-13, Berlin: <a class="moz-txt-link-freetext" href="http://asipto.com/u/kat">http://asipto.com/u/kat</a>
<a class="moz-txt-link-freetext" href="http://linkedin.com/in/miconda">http://linkedin.com/in/miconda</a> -- <a class="moz-txt-link-freetext" href="http://twitter.com/miconda">http://twitter.com/miconda</a>


_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
          </blockquote>
          <pre wrap="">

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
        </blockquote>
        <pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>

</pre>
      </blockquote>
      <pre wrap="">

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users at lists.sip-router.org">sr-users at lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>

</pre>
    </blockquote>
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--------------010100070407070903000403--


From bogus@does.not.exist.com  Fri Jun  3 15:33:31 2011
From: bogus@does.not.exist.com ()
Date: Fri, 03 Jun 2011 13:33:31 -0000
Subject: No subject
Message-ID: <mailman.11.1311886593.2669.sr-users@lists.sip-router.org>

--=20
I=C3=B1aki Baz Castillo
<ibc at aliax.net>