[asterisk-bugs] [JIRA] Commented: (ASTERISK-18220) MixMonitor stops recording during attended Transfer

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-18220?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=196262#comment-196262 ] 

Matt Jordan commented on ASTERISK-18220:
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I think we talked a bit about this on the -users list.  If nothing else, it feels like audiohooks should probably be documented on the Asterisk wiki a bit more, including how and when to use them with MixMonitor.

> MixMonitor stops recording during attended Transfer
> ---------------------------------------------------
>
>                 Key: ASTERISK-18220
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18220
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_mixmonitor
>    Affects Versions: 1.8.3
>         Environment: CentOs 5.5 & 5.6
>            Reporter: Ishfaq Malik
>
> I have the following scenario...
> SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called)
> Extension A puts call on hold and calls extension B
> Extension A then does an attended transfer of incoming call to extension B
> I'm finding that the recording only lasts up to the point that the transfer is made.
> Here is a section of the logs for such a call
> [2011-08-02 13:47:13] VERBOSE[6475] rtp_engine.c:     -- Locally bridging SIP/A-00000049 and SIP/B-0000004a
> [2011-08-02 13:47:20] VERBOSE[6475] rtp_engine.c:     -- Locally bridging SIP/inbound-00000047 and SIP/B-0000004a
> [2011-08-02 13:47:20] VERBOSE[6463] pbx.c:   == Spawn extension (inbound, s, 4) exited non-zero on 'SIP/A-00000049<ZOMBIE>'
> [2011-08-02 13:47:20] VERBOSE[6464] app_mixmonitor.c:   == MixMonitor close filestream
> [2011-08-02 13:47:26] VERBOSE[6475] app_macro.c:   == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/inbound-00000047' in macro 'stdexten'
> [2011-08-02 13:47:26] VERBOSE[6475] pbx.c:   == Spawn extension (local, B, 1) exited non-zero on 'SIP/inbound-00000047'
> [2011-08-02 13:47:26] VERBOSE[6464] app_mixmonitor.c:   == End MixMonitor Recording SIP/inbound-00000047
> Obviously, I've obscured some of the more sensitive details in there
> The thing to notice here though is that MixMonitor closes the filestream when I hit the transfer button but actually Ends the recording 6 seconds later when the whole call was ended.
> This seems like inconsistent behaviour and more like an unintentional consequence of changes rather than intended behaviour, i.e. why would you close the filestream yet not end the recording?
> Also, this is a departure from the behaviour in 1.4 where the entire call would be recorded.

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