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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:
<blockquote
cite="mid:4263BC1293C60E4982E3DFFFF19FCDB1A94777@UK-MAIL-STOR-1.thebigword.com"
type="cite">
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<div>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.</div>
<div> </div>
<div>Thanks</div>
<div> </div>
<div>Lee</div>
<div> </div>
</span></font>
<pre>thebigword Holdings Limited. Registered Office: Link Up House, Ring Road, Lower Wortley, Leeds, UK, LS12 6AB. Registered in England & Wales No. 05551907
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<br>
<br>
<pre class="moz-signature" cols="72">--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"</pre>
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