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Today's Topics:

   1. Re: Dialog module with 2 servers and 2    separate    databases.
(Daniel-Constantin Mierla)
   2. First public release of sip:provider Community Edition
(Andreas Granig)
   3. First public release of sip:provider Community Edition
(Andreas Granig)
   4. Re: Crash (michel freiha)

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

Message: 1
Date: Mon, 13 Dec 2010 15:12:13 +0100
From: Daniel-Constantin Mierla <miconda@gmail.com>
Subject: Re: [SR-Users] Dialog module with 2 servers and 2    separate
    databases.
To: "Pan B. Christensen" <pan@ibidium.no>
Cc: sr-users@lists.sip-router.org
Message-ID: <4D0629BD.8050501@gmail.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

Hello,

one option you can do is to write to db immediately when a call is
active (see dialog module parameters) and do a query to the other server
database in addition to counting the local instance active calls.

Another one, different, is to use memcache for a communication system
between two or more instances.

Cheers,
Daniel

On 12/13/10 2:37 PM, Pan B. Christensen wrote:
> Thanks for your reply, Daniel.
> The purpose is to do busy forwarding without querying the client when
> the user already has >= X active calls. X will normally be 1 (call
> waiting inactive) or 2 (call waiting active). Advanced users may
> possibly set a higher value. Counting the number of calls on
> the server and doing busy forwarding based on that rather than waiting
> for a "486 Busy here" from the client has several advantages.
> Currently, I've written code to do this with the dispatcher module,
> and it's working great with only one server. Here's a code snippet:
>    $var(dlg_busy) = 0;
>    get_profile_size("busy", "$avp(s:uid)", "$var(dlg_busy)");
>    if ( $var(dlg_busy) >= $avp(s:busy_level) ) {
>    if ($avp(s:cfb_status) == "on") {
>    $rU = $avp(s:cfb_number);
>    xlog("L_INFO", "-------------------- $avp(s:uid) has
> $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding
> on busy to $rU --------------------\n");
>    route(10);
>    }
>    ...
>    }
> Based on your reply, I guess one way to solve this would be to write
> the get_profile_size function in sqlops, query the two dialog
> databases and add the numbers. This would still require the customer
> to change their database design. Is there an easier or better way to
> do this?
> I also wote code to do busy forwarding if the client replies with 486
> (do not disturb activated), 603 (call rejected) etc.
> This code works for normal busy forwarding if Polycom is set to 1 call
> per line key (default is 8). We'll then have to provision the
> $avp(s:busy_level) variable to the clients instead of handling it
> server-side. If a user now wants to change the setting, he'll have to
> reboot his phone after doing so. Changing the
> reg.x.callsPerLineKey setting in the phone also limits the number of
> outgoing calls the user can make. We'll also have to make code for all
> the other hardphones the customer is planning to use plus make guides
> on how to change the setting for all kinds of softphones. We want to
> avoid all this.
> With kind regards,
> Pan
>
>    ----- Original Message -----
>    *From:* Daniel-Constantin Mierla <mailto:miconda@gmail.com>
>    *To:* Pan B. Christensen <mailto:pan@ibidium.no>
>    *Cc:* sr-users@lists.sip-router.org
>    <mailto:sr-users@lists.sip-router.org>
>    *Sent:* Monday, December 13, 2010 12:26 PM
>    *Subject:* Re: [SR-Users] Dialog module with 2 servers and 2
>    separate databases.
>
>
>
>    On 12/10/10 2:17 PM, Pan B. Christensen wrote:
>>    Hello,
>>    My customer has the following database design.
>>    Voip server 1 talks to SQL server 1.
>>    Voip server 2 talks to SQL server 2.
>>    Voip 1 and Voip 2 are load-balanced.
>>    Each SQL server has two databases. Database 1 contains
>>    semi-static data like call forwarding properties for users and is
>>    read-only. This is replicated from a third SQL server which the
>>    web interface writes to. Database 2 is read/write, is not
>>    replicated and contains data that is updated frequently like user
>>    location and now dialog info.
>>    Voip server 1 is not allowed to talk to SQL server 2 and vice versa.
>>    I'm using forward() to send authenticated REGISTERs to the other
>>    server so that it'll write this to RAM and its own SQL server.
>>    Thus, both servers are aware of clients authenticated and
>>    registered by the other server.
>>    How can I make both servers be aware of active calls on the other
>>    server?
>    what is the purpose?
>
>    Practically, it is not possible to track a call in two instances,
>    because, unlike registration where is just a storage of mappings
>    between contact and aor, call states of dialog module involve more
>    processing logic, including timeouts and sending BYEs.
>
>    Cheers,
>    Daniel
>
>    --
>    Daniel-Constantin Mierla
>    Kamailio (OpenSER) Advanced Training
>    Jan 24-26, 2011, Irvine, CA, USA
>    http://www.asipto.com
>
>    ------------------------------------------------------------------------
>
>    No virus found in this incoming message.
>    Checked by AVG - www.avg.com
>    Version: 8.5.449 / Virus Database: 271.1.1/3312 - Release Date:
>    12/12/10 19:34:00
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com

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Message: 2
Date: Mon, 13 Dec 2010 15:42:43 +0100
From: Andreas Granig <agranig@sipwise.com>
Subject: [SR-Users] First public release of sip:provider Community
    Edition
To: sr-users@lists.sip-router.org
Message-ID: <4D0630E3.7000204@sipwise.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi all,

We at Sipwise are excited to announce the first public release of the
sip:provider Community Edition (http://www.sipwise.com/products/spce/).
It is a fully open-source SIP based Class5 VoIP soft-switch, providing
every component an operator needs to offer VoIP services. It comes as a
communication platform leveraging the capabilities of Kamailio, SEMS and
Asterisk, complemented by our own open-sourced building blocks to
provide consistent and easy-to-use provisioning, billing and
configuration maintenance. The different parts are carefully integrated
with each other to form a fully featured VoIP soft-switch.

The platform will make it much easier for new Kamailio users to get
started with VoIP, and will provide missing parts in the open-source
VoIP eco-system for more experienced users.

Please check
http://www.sipwise.com/news/announcements/spce-first-release/ for more
information on this release.

Have fun playing with it, we hope it's as useful to you as it is for us.
Andreas

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Message: 3
Date: Mon, 13 Dec 2010 15:45:36 +0100
From: Andreas Granig <agranig@sipwise.com>
Subject: [SR-Users] First public release of sip:provider Community
    Edition
To: kamailio <sr-users@lists.sip-router.org>
Message-ID: <4D063190.4000902@sipwise.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi all,

We at Sipwise are excited to announce the first public release of the
sip:provider Community Edition (http://www.sipwise.com/products/spce/).
It is a fully open-source SIP based Class5 VoIP soft-switch, providing
every component an operator needs to offer VoIP services. It comes as a
communication platform leveraging the capabilities of Kamailio, SEMS and
Asterisk, complemented by our own open-sourced building blocks to
provide consistent and easy-to-use provisioning, billing and
configuration maintenance. The different parts are carefully integrated
with each other to form a fully featured VoIP soft-switch.

Please check
http://www.sipwise.com/news/announcements/spce-first-release/ for more
information on this release.

Have fun playing with it, we hope it's as useful to you as it is for us.
Andreas

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Message: 4
Date: Mon, 13 Dec 2010 16:48:02 +0200
From: michel freiha <michofr@gmail.com>
Subject: Re: [SR-Users] Crash
To: Daniel-Constantin Mierla <miconda@gmail.com>
Cc: users@lists.kamailio.org
Message-ID:
    <AANLkTi=5-JpFG3q=KQM-LK-dpKGYn03w1yYwLJpPfY=N@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hello Daniel,

We are not facing any database problem even we are not using DNS anywhere in
Kamailio config...The only thing is that the debug level is 2 and syslog was
not configured asynchronous...Do you think if we configure it as
asynchronous, our issue will be solved?

Regards

On Mon, Dec 13, 2010 at 1:08 PM, Daniel-Constantin Mierla <miconda@gmail.com
> wrote:

> Hello,
>
>
> On 12/10/10 4:15 PM, michel freiha wrote:
>
>> Hello Sir,
>>
>> The crash issue has been successfully fixed after using GIT for the
>> version 3.1.0.
>>
> thanks for reporting back.
>
>
> Now we have another problem..>When the number of registered users exceeded
>> 2500 concurrent registered users, the kamailio stuck and each call will take
>> up to 1 minute to be established
>>
>> Any comment on that?
>>
> Do you have high debug level? If yes, is your syslog configured
> asynchronously?
>
> Other than that, you can use benchmark module to spot which of your config
> actions takes so long to execute. Normally, such cases can happen when you
> have queries to slow database or dns servers.
>
> Cheers,
> Daniel
>
>
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Jan 24-26, 2011, Irvine, CA, USA
> http://www.asipto.com
>
>
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