[asterisk-users] Recording with MixMonitor and AGI

[Bharat Lalcheta]

As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension.  As your channel already hangup, it can not
run on AGI.

Hope it will help you.

Regards,

Bharat Lalcheta

On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
<henrik.westerberg at ain.se> wrote:
> Hi,
>
> I am developing a call recording application on Asterisk 11.2 and have this
> configuration in my dialplan:
>
> [macro-ccdev2-rec]
> exten => s,1,MixMonitor(${ARG1},b)
>
> [outgoing-originate]
> exten => _X.,1,NoOp(Will send call to ${EXTEN})
> exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z)
>
> [outgoing-originate-rec]
> exten =>
> h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})
>
> exten => _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID},
> CC_FILENAME is ${CC_FILENAME})
> exten => _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
>
> If I want to make a recorded server callout from 077777777 to 0888888888 I
> then originate a call via AMI to Local/077777777 at outgoing-originate with
> context set to outgoing-originate-rec and extension to 0888888888.
> The result will be something like this:
>
>     -- Executing [s at macro-ccdev2-rec:1]
> MixMonitor("SIP/upps-ccm-tq01-0000003f", "cbrec-15605.wav,b") in new stack
>   == Begin MixMonitor Recording SIP/upps-ccm-tq01-0000003f
>     -- Executing [h at outgoing-originate-rec:1]
> AGI("SIP/upps-ccm-tq01-0000003e",
> "agi://l4574/ajpbxtest.agi?path=uploadrec&callid=15605") in new stack
>     -- <SIP/upps-ccm-tq01-0000003e>AGI Script
> agi://localhost/ajpbxtest.agi?path=uploadrec&callid=15605 completed,
> returning 0
>     -- Executing [h at outgoing-originate-rec-dev2:1]
> AGI("SIP/upps-ccm-tq01-0000003f",
> "agi://4574/ajpbxtest.agi?path=uploadrec&callid=") in new stack
>     -- <SIP/upps-ccm-tq01-0000003f>AGI Script
> agi://localhost/ajpbxtest.agi?path=uploadrec&callid= completed, returning 0
>   == MixMonitor close filestream (mixed)
>   == End MixMonitor Recording SIP/upps-ccm-tq01-0000003f
>
> Unfortunately I get two different calls to the h extension, but this I can
> cope with. The one without called is not interesting.
> The uploading will fail since the MixMonitor is still on when I try to
> upload the file. The file will not have a duration. It works when I schedule
> the uploading a while after from my agi application but I would rather not
> rely on a timeout.
>
> When I tried to run StopMixMonitor before the Agi call in the h extension,
> the first call fail and I never get any uploading with callid.
>
>     -- Executing [s at macro-ccdev2-rec:1]
> MixMonitor("SIP/upps-ccm-tq01-00000043", "cbrec-15607.wav,b") in new stack
>   == Begin MixMonitor Recording SIP/upps-ccm-tq01-00000043
>     -- Executing [h at outgoing-originate-rec-dev2:1]
> StopMixMonitor("SIP/upps-ccm-tq01-00000042", "") in new stack
>   == Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on
> 'SIP/upps-ccm-tq01-00000042'
>     -- Executing [h at outgoing-originate-rec-dev2:1]
> StopMixMonitor("SIP/upps-ccm-tq01-00000043", "") in new stack
>   == MixMonitor close filestream (mixed)
>     -- Executing [h at outgoing-originate-rec-dev2:2]
> AGI("SIP/upps-ccm-tq01-00000043",
> "agi://localhost/ajpbxtest.agi?path=uploadrec&callid=") in new stack
>
> Am I missing something here? I also looked at the possibility to specify a
> command to execute when MixMonitor stops but I would rather handle the file
> uploading in my agi application.
>
> I also have another case: I want to dial out a call and record it. It will
> be a "oneway-call" from the server to a mobile. Do I need to get AGI-control
> of it and record with an AGI command or how can I hack it directly in the
> dial plan using MixMonitor?
>
> Best Regards,
> Henrik
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Bharat Lalcheta