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<font face="monospace">Hi Antonis,<br>
<br>
I guess you should move the enagaing of the rtpengine (for the
INVITE time) in the branch route, so it can do different SDP
settings according to the destination of that branch.<br>
<br>
How you do it now will set the SDP in INVITE only in relation to
the first branch (so WAN) and reuse it for the next branches too.<br>
<br>
Best regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 2021
<a class="moz-txt-link-freetext" href="https://opensips.org/training/OpenSIPS_eBootcamp_2021/">https://opensips.org/training/OpenSIPS_eBootcamp_2021/</a></pre>
<div class="moz-cite-prefix">On 9/27/21 3:48 AM, Antonis Psaras
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:01f401d7b339$743a6210$5caf2630$@microbase.gr">
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<div class="WordSection1">
<p class="MsoNormal"><span style="color:#595959">Hello Team<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have an OpenSIPs with multi home (WAN /
LAN) connected to an Asterisk (LAN). The problem I have is the
following.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">A call is coming from Asterisk to OpenSIPs
LAN interface for a user registered on OpenSIPs. That user has
call forwarding enabled and OpenSIPs receives a 302. <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">That request is handled as follows<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">failure_route[failure]<o:p></o:p></p>
<p class="MsoNormal">{<o:p></o:p></p>
<p class="MsoNormal"> if (t_check_status("(301)|(302)"))<o:p></o:p></p>
<p class="MsoNormal"> {<o:p></o:p></p>
<p class="MsoNormal"> get_redirects("1:1");<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> uac_replace_from("","$tu");<o:p></o:p></p>
<p class="MsoNormal"> uac_replace_to("","$ru");<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> if (!ds_select_dst("1",
"0"))<o:p></o:p></p>
<p class="MsoNormal"> {<o:p></o:p></p>
<p class="MsoNormal">
send_reply("500","Unable to route");<o:p></o:p></p>
<p class="MsoNormal"> exit;<o:p></o:p></p>
<p class="MsoNormal"> }<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> t_relay();<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> }<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> rtpengine_delete();<o:p></o:p></p>
<p class="MsoNormal">}<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">A new INVITE is generated from OpenSIPs
towards Asterisk but SDP negotiated is the initial (OpenSIPs
to Client) with the WAN IP.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Is there a way to correct that?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you in advance for your support.<o:p></o:p></p>
</div>
<br>
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<pre class="moz-quote-pre" wrap="">_______________________________________________
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</pre>
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