[asterisk-bugs] [JIRA] (ASTERISK-26168) app_mixmonitor stop during a call

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26168?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231269#comment-231269 ] 

Asterisk Team commented on ASTERISK-26168:
------------------------------------------

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].

> app_mixmonitor stop during a call
> ---------------------------------
>
>                 Key: ASTERISK-26168
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26168
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_mixmonitor
>    Affects Versions: 13.7.0
>            Reporter: Hervé Jacquemin
>            Severity: Minor
>
> 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



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