<div class="gmail_quote">On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina <span dir="ltr"><<a href="mailto:mmolina@millenium.com.co">mmolina@millenium.com.co</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
ast guy escribió:<br>
<div><div></div><div class="h5">> Hi,<br>
> I am using codec g729 on two asterisk machines, but when call is<br>
> forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1<br>
> outputs following error and there is no audio. Also the IVRs being<br>
> played have choppy voice.<br>
><br>
> "Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c<br>
> = '')"<br>
><br>
> It is running fine when codec gsm is in RTP traffic.<br>
><br>
> Also I have another server 3 which is also running g729, call from<br>
> server 3 to server 2 is established but still choppy voice. Earlier I<br>
> integrated server 3 to server 1 and it was a smooth run.<br>
><br>
> Any idea what could be the possible reasons!<br>
><br>
> /ag<br>
</div></div>Please provide the asterisk version and g729 codec that is installed on<br>
each server, so people can have a clue of what's happening. Maybe could<br>
be a known bug or something.<br>
<br>
Cheers,<br>
<font color="#888888"><br>
--<br>
Ing. Miguel Molina<br>
Grupo de Tecnología<br>
Millenium Phone Center<br>
</font><div><div></div><div class="h5"><br>
</div></div></blockquote></div><br><br>I am running Asterisk 1.2.13. I need to look for the actual source from where I got the codec.<br><br><br>/ag<br>