[OpenSIPS-Users] Renegotiation

Bogdan-Andrei Iancu bogdan at opensips.org
Wed Mar 5 18:42:47 CET 2014


Hello Jorge,

You need to use the failure route - 
http://www.opensips.org/Documentation/Script-Routes-1-10#toc3 .

Use the t_on_failure() function (from TM module) to arm a failure route 
before sending the INVITE out (before the t_relay()). The failure route 
will be triggered when receiving the negative reply - use 
t_check_status() 
[http://www.opensips.org/html/docs/modules/1.10.x/tm.html#id294673] to 
check the reply code and if 415, simply change the RURI and send the 
INVITE out again.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 04.03.2014 13:13, Jorge Ortea wrote:
> Hi all,
>
> Anyone know how to do this?
>
> Thanks.
> Regards.
>
>
> ------------------------------------------------------------------------
> From: darham at hotmail.com
> To: stefano.pisani at omnianet.it; users at lists.opensips.org
> Date: Fri, 28 Feb 2014 13:12:16 +0100
> Subject: Re: [OpenSIPS-Users] Renegotiation
>
> Hi Stefano,
>
> How I can do that?
>
> Very thanks.
> Regards.
>
>
> ------------------------------------------------------------------------
> Date: Thu, 27 Feb 2014 17:04:29 +0100
> From: stefano.pisani at omnianet.it
> To: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] Renegotiation
>
> You can trap the 415 response from the called peer and send the call 
> through asterisk to get transcoding.
>
> Il 27/02/2014 16.51, Jorge Ortea ha scritto:
>
>     Hi all,
>
>     I have a scenario with OpenSIPS 1.8 and Asterisks 1.4.   Proxy SIP
>     has two ways to manage a call, the first is B2BUA and second is be
>     relay between UAC and Asterisk.
>
>     I have a problem, when OpenSIPS works as B2BUA and both UAC can't
>     negotiate codec then this call failed. I would like restart this
>     same call at second way (with Asterisk). Is that possible?
>
>     Thanks.
>     Regards.
>
>
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