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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Thanks for the help Bogdan. I will do some tests and see if I can figure it.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Regards<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Lee<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext"> Bogdan-Andrei Iancu [mailto:bogdan@opensips.org]
<br>
<b>Sent:</b> 28 December 2011 13:08<br>
<b>To:</b> OpenSIPS users mailling list<br>
<b>Cc:</b> Lee Archer<br>
<b>Subject:</b> Re: [OpenSIPS-Users] B2BUA + RTPproxy + Asterisk direct media<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hello Lee,<br>
<br>
Asterisk is doing the "direct media" by firing some re-INVITEs after the call is up in order to exchange the media IPs of the the end points.<br>
<br>
So, if this does not work, most probably you do not correctly handle the re-INVITEs in opensips, like you are no forcing again rtpproxy for re-INVITEs.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
On 12/28/2011 12:26 PM, Lee Archer wrote: <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Hi all, I wonder if someone can help me. I have a system where I use the B2B module and RTPproxy for inbound calls but once answered the call might jump between Asterisk
servers depending on what service is required. I would like to use the Asterisk direct media option for SIP calls but when enabled the server is trying to talk to the SIP providers RTP gateway instead of my RTPproxy instance. I’ve made changes to the RTPproxy
configuration but I’m wondering if anyone else uses direct media with RTPproxy and can point me in the right direction config wise.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> <o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Thanks<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> <o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Lee<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> <o:p></o:p></span></p>
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<pre>thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower Wortley, Leeds, UK, LS12 6AB. Registered in England & Wales No. 05551907<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Users mailing list<o:p></o:p></pre>
<pre><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><o:p></o:p></pre>
<pre><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre>
<p class="MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre>
<pre>Bogdan-Andrei Iancu<o:p></o:p></pre>
<pre>OpenSIPS Founder and Developer<o:p></o:p></pre>
<pre>OpenSIPS solutions and "know-how"<o:p></o:p></pre>
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<pre>thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower Wortley, Leeds, UK, LS12 6AB. Registered in England & Wales No. 05551907