[Asterisk-Users] Codecs problem
That's a call to pstn Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that there is no match and give me an error :( Any idea? Kind regards, Olivier 9 headers, 11 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 82.146.123.246:38098 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs problem
I've found that happens when one version of asterisk is 1.2 and the other end is running 1.0.9 and you are connecting over IAX2. If you bridge the two servers with SIP it will be fine. -bill On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote: That's a call to pstn Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that there is no match and give me an error :( Any idea? Kind regards, Olivier 9 headers, 11 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 82.146.123.246:38098 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Codecs problem
Unfortunately, we are on sip :( Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de William Lloyd Envoyé : mercredi 9 novembre 2005 18:12 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Codecs problem I've found that happens when one version of asterisk is 1.2 and the other end is running 1.0.9 and you are connecting over IAX2. If you bridge the two servers with SIP it will be fine. -bill On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote: That's a call to pstn Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that there is no match and give me an error :( Any idea? Kind regards, Olivier 9 headers, 11 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 82.146.123.246:38098 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs Problem?
Hello, I have a following setup: IP phone (Cisco/Skinny) - * - NAT -- NAT - * - PSTN Everything is perfect when i'm using it from right to left. From left to right however, there is no voice, although the calls are being placed. I played around with codeces but no change. Does anybody know, what I possibly am doing wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users