[asterisk-users] Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another extensions, while this outside call is waiting with music, the another extension call hangs up suddenly, and the call is back to the outside call suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel: SIP/219-081d4d60 Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels SIP/219-081d4d60 and Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on channel 1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with 0 conference users 15 13:32:55 VERBOSE[27850] logger.c: -- Hungup 'Zap/1-1' Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 15 13:32:55 VERBOSE[27850] logger.c: == Spawn extension This may be the cause: Didn't get a frame from channel... I googled. It is recommended to disable busydetect, but no solution. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sergio de los Santos ssantos @ hispasec.com Hispasec Sistemas S.L 902 161 025 29590 Málaga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another extensions, while this outside call is waiting with music, the another extension call hangs up suddenly, and the call is back to the outside call suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel: SIP/219-081d4d60 Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels SIP/219-081d4d60 and Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on channel 1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with 0 conference users 15 13:32:55 VERBOSE[27850] logger.c: -- Hungup 'Zap/1-1' Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 15 13:32:55 VERBOSE[27850] logger.c: == Spawn extension This may be the cause: Didn't get a frame from channel... I googled. It is recommended to disable busydetect, but no solution. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] didn't get a frame from channel
This is an excerpt from the log file, My problem is that randomly, 1out of 3 or 1 out of 2, some calls are not going out and this is the message in the log file, The device that should provide the frame is a Sipura 3000 which has its FXO providing outside connectivity, 24185 Sep 16 10:35:40 DEBUG[17604]: Didn't get a frame from channel: SIP/200-b635 24186 Sep 16 10:35:40 DEBUG[17604]: Bridge stops bridging channels SIP/200-b635 and SIP/pstn_1-a971 24187 Sep 16 10:35:40 DEBUG[17604]: update_user_counter(ww9863038) - decrement outUse counter 24188 Sep 16 10:35:40 DEBUG[17604]: ww9863038 is not a local user 24189 Sep 16 10:35:40 DEBUG[17604]: Exiting with DIALSTATUS=ANSWER. I can provide the whole call log but it's many lines long. Any clue on where to look for anything? Thanks Andres ___ --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
