Hi,<br> I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice.<br>
<br>"Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = '')"<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 server 3 to server 2 is established but still choppy voice. Earlier I 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>