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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
