Summary:	ASTERISK-26168: --- SOLVED --- Please close app_mixmonitor stop during a call
Reporter:	Hervé Jacquemin (hjacquemin)	Labels:
Date Opened:	2016-07-01 04:23:18	Date Closed:	2016-07-01 04:46:17
Priority:	Minor	Regression?	Yes
Status:	Closed/Complete	Components:	Applications/app_mixmonitor
Versions:	13.7.0	Frequency of Occurrence	Constant
Related Issues:
Environment:		Attachments:
Description:	---- EDIT ---- Please close, customer made job in my back and breake it with crontab... I faced an issue with app_mixmonitor in Asterisk 13.7.0. When I tried to use Online Call Recording, everything seems to be good (at the logs level) but in fact the audio file is not complete. Lets check the logs: [Jun 30 15:34:55] VERBOSE[399][C-00002e90] app_dial.c: Called SIP/3StarsNet/xxxxxxxxx [Jun 30 15:34:57] VERBOSE[399][C-00002e90] app_dial.c: SIP/3StarsNet-0000191e is making progress passing it to SIP/wn8dca-0000191d [Jun 30 15:34:57] VERBOSE[399][C-00002e90] res_rtp_asterisk.c: 0x7f777c848f30 -- Probation passed - setting RTP source address to 188.66.8.26:14772 [Jun 30 15:34:57] VERBOSE[399][C-00002e90] res_rtp_asterisk.c: 0x7f77c00b71e0 -- Probation passed - setting RTP source address to 192.50.1.220:11782 [Jun 30 15:34:59] VERBOSE[399][C-00002e90] app_dial.c: SIP/3StarsNet-0000191e is ringing [Jun 30 15:35:08] VERBOSE[399][C-00002e90] app_dial.c: SIP/3StarsNet-0000191e answered SIP/wn8dca-0000191d [Jun 30 15:35:08] VERBOSE[914][C-00002e90] bridge_channel.c: Channel SIP/3StarsNet-0000191e joined 'simple_bridge' basic-bridge <9cc299d9-73ae-4e33-a0ad-1c0f530a2911> [Jun 30 15:35:08] VERBOSE[399][C-00002e90] bridge_channel.c: Channel SIP/wn8dca-0000191d joined 'simple_bridge' basic-bridge <9cc299d9-73ae-4e33-a0ad-1c0f530a2911> [Jun 30 15:35:09] DTMF[399][C-00002e90] channel.c: DTMF end '*' received on SIP/wn8dca-0000191d, duration 300 ms [Jun 30 15:35:09] DTMF[399][C-00002e90] channel.c: DTMF begin emulation of '*' with duration 300 queued on SIP/wn8dca-0000191d [Jun 30 15:35:09] DTMF[399][C-00002e90] channel.c: DTMF end emulation of '*' queued on SIP/wn8dca-0000191d [Jun 30 15:35:10] DTMF[399][C-00002e90] channel.c: DTMF end '3' received on SIP/wn8dca-0000191d, duration 300 ms [Jun 30 15:35:10] DTMF[399][C-00002e90] channel.c: DTMF begin emulation of '3' with duration 300 queued on SIP/wn8dca-0000191d [Jun 30 15:35:10] DTMF[399][C-00002e90] channel.c: DTMF end emulation of '3' queued on SIP/wn8dca-0000191d *+[Jun 30 15:35:10] VERBOSE[399][C-00002e90] bridge_builtin_features.c: AutoMixMonitor used to record call. Filename: auto-1467293710-023322130-089700454.wav+* [Jun 30 15:35:10] VERBOSE[1006][C-00002e90] app_mixmonitor.c: Begin MixMonitor Recording SIP/3StarsNet-0000191e [Jun 30 15:36:33] VERBOSE[399][C-00002e90] bridge_channel.c: Channel SIP/wn8dca-0000191d left 'simple_bridge' basic-bridge <9cc299d9-73ae-4e33-a0ad-1c0f530a2911> [Jun 30 15:36:33] VERBOSE[399][C-00002e90] pbx.c: Spawn extension (outcall, dial, 7) exited non-zero on 'SIP/wn8dca-0000191d' [Jun 30 15:36:33] VERBOSE[914][C-00002e90] bridge_channel.c: Channel SIP/3StarsNet-0000191e left 'simple_bridge' basic-bridge <9cc299d9-73ae-4e33-a0ad-1c0f530a2911> +*[Jun 30 15:36:33] VERBOSE[1006][C-00002e90] app_mixmonitor.c: MixMonitor close filestream (mixed)*+ [Jun 30 15:36:33] VERBOSE[1006][C-00002e90] app_mixmonitor.c: End MixMonitor Recording SIP/3StarsNet-0000191e The recording seems to be actif until the end of the calls (+/- 1min13sec), but when I download the audio file is only 51sec. The Automixmon seems to stop at a certain moment. If one of you has an idea to solve this... Regards, Hervé Jacquemin
Comments:	By: Asterisk Team (asteriskteam) 2016-07-01 04:23:19.012-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].