[asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
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 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 ___ -- 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
Re: [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
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 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
Re: [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina mmol...@millenium.com.cowrote: 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 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 ___ -- 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
