[asterisk-users] trouble recording calls

[BJ Weschke]

On 4/9/07, ahester <ahester at architel.com> wrote:
> Hi all,
>
> I am having the following trouble with recording calls:
> When calls come into the support line did number, the call starts to
> record on the first queue, but appears to hang up when the call actually
> connects to the engineer (ie I see "got hangup request" on the cli and
> then mixmonitor ends.)  I am guessing this has to do with the announce
> file that is played to the engineer before the call is connected.  It
> seems that if the call rolls to the next queue because of timeout,
> asterisk doesn't even try to record it. (I don't see any mixmonitor on
> the cli for the next queue).
>
> I would appreciate any help with this.  I have to have all calls
> recorded and I have to do announcements so that the callee knows how to
> answer the phone.
>
> Thanks,
> Andy
>
>
> The configs are as below:
>
> >From extensions.conf:
>
> #after various menu stuff, send to support
> exten => 214xxxxxxx,13,SetGlobalVar(ORIGIN=support)
> exten => 214xxxxxxx,14,Queue(support1|tr|||10)
> exten => 214xxxxxxx,15,Queue(support2|tr|||)
>
> #dial command for sip extensions that are in the queues
> exten =>
> _72XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
> exten => _72XXX,2,Dial(SIP/${EXTEN})
> exten =>
> _73XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
> exten => _73XXX,2,Dial(SIP/${EXTEN})
>
>
> queues from queues.conf:
>
> [support1]
> ; Support call queue
> announce = 16
> strategy = rrmemory
> timeout = 15
> retry = none
> wrapuptime=15
> announce-frequency = 0
> joinempty = no
> leavewhenempty = yes
> member => Agent/2008
> member => Agent/2009
> member => Agent/2014
> member => Agent/2015
> member => Agent/2017
> member => Agent/2018
> member => Agent/2019
> member => Agent/3520
> member => Agent/3521
> member => Agent/3522
> member => Agent/3524
> member => Agent/3529
>
> [support2]
> ; Support2 call queue
> announce = 16
> strategy = ringall
> announce-frequency = 0
> ; Added below for testing because the second queue was not even trying
> to record
> ; according to the asterisk console (still doesn't)
> Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
> monitor-format = wav
> monitor-join = yes
> joinempty = yes
> member => SIP/72008
> member => SIP/72009


 You cannot use the MixMonitor app on its own in a callback scenario
because, as you've already discovered, MixMonitor senses the call
transition between the time the agent answers and the calls is then
bridged with the waiting caller and still stop recording.

 To fix this, in the 1.4 version of app_queue, there's a
"monitor-type=MixMonitor" parameter which will use the MixMonitor
appropriately natively in app_queue instead of Monitor.

 BJ

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