On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina <[email protected]>wrote:
> ast guy escribió: > > 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 > Please provide the asterisk version and g729 codec that is installed on > each server, so people can have a clue of what's happening. Maybe could > be a known bug or something. > > Cheers, > > -- > Ing. Miguel Molina > Grupo de Tecnología > Millenium Phone Center > > I am running Asterisk 1.2.13. I need to look for the actual source from where I got the codec. /ag
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