Hi, 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.
"Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = '')" It is running fine when codec gsm is in RTP traffic. 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. Any idea what could be the possible reasons! /ag
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