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: dar...@hotmail.com
To: stefano.pis...@omnianet.it; users@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.pis...@omnianet.it
To: users@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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