[asterisk-users] Why did it Hangup?

Wed May 9 00:08:34 CDT 2012

Hi Shahid,

I am in favor of asterisk for what it is doing to your call. When you send
an AMI event like the one you wrote it sends the A party invite right away
w/o going into any context/extension. As soon as the A-party answers the
call Asterisk manager connects/lands A-channel to the test context
extension 210.

Your monitor command is recording I/O streams of b-party here. which is
nothing except SendDTMF (silence I'd say in terms of audio) - using
MixMonitor, as Danny said, will record both A; and B-party channel audio.

Or the other possibility is that instead of directly making out call to
A-party i.e SIP/447XXXXXXX at vpsprovider, SET some variables and originate
call to Local channel i.e Local/447XXXXXXX at my-demo-context. And in that
context use Monitor() before DIAL()

[my-demo-context]
exten => _447XXXXXXX,1,NOOP(--- I'm going to Dial ${EXTEN} ---)
same => n,Monitor(blah-blah-blah filename)
same => n,DIAL(SIP/${EXTEN}@vpsprovider,,gFD( wwww2w3w))
same => n,StopMonitor()
same => n,Hangup()

NOTE: You should not put and ANSWER() in this context/extension as that
will lead to immediate connection to the B-party context^exten^prio
(test,210,1 in your case)

The only difference this approach would make is that it'll record the
ringing sounds/busy tones or any other carrier error messages played on
your desired Outbound number.Whereas in your case only successful calls
will be recorded.

Looking at your scripts I'm having a feeling that you are trying to make an
Auto-IVR-tester sort of thing - pressing DTMFs on an IVR and recording
audio to see if your desired IVR has played or not !!

Regards,
Sammy.

On Wed, May 9, 2012 at 1:41 AM, Danny Nicholas <danny at debsinc.com> wrote:

> Try MixMonitor instead of Monitor.  I think it is more robust.****
>
> ** **
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Shahid H
> *Sent:* Tuesday, May 08, 2012 3:39 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Why did it Hangup?****
>
> ** **
>
> I have tried that and that did not fixed the problem,****
>
> ** **
>
> However, I have added this in the dialplan:****
>
> ** **
>
> exten => 210,n,Wait(60)****
>
> ** **
>
> That will hangup the call after 60 seconds...****
>
> ** **
>
> That is fine by me but now Monitor() dont even work now... it does not
> record a call...?****
>
> ** **
>
> ** **
>
> ** **
>
> On Tue, May 8, 2012 at 9:22 PM, Danny Nicholas <danny at debsinc.com> wrote:*
> ***
>
> Since you are Originating the call, the hangup command isn’t needed.
> Remove and reload.****
>
>  ****
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Shahid H
> *Sent:* Tuesday, May 08, 2012 3:20 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Why did it Hangup?****
>
>  ****
>
> No, that 'timeout' option is when I don't answer the call.****
>
>  ****
>
> My problem is when I DO answer the call, it get disconnected right away.**
> **
>
>  ****
>
> Yes hangup() get executed right away when I answer the call.****
>
>  ****
>
>  ****
>
> On Tue, May 8, 2012 at 9:17 PM, Danny Nicholas <danny at debsinc.com> wrote:*
> ***
>
> It is likely the 60 second timeout you are providing.  Or it could be the
> hangup() command in the 210 context.****
>
>  ****
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Shahid H
> *Sent:* Tuesday, May 08, 2012 3:11 PM
> *To:* asterisk-users at lists.digium.com
> *Subject:* [asterisk-users] Why did it Hangup?****
>
>  ****
>
> I am learning how to use AMI and I am having 1 problem.. When I make a
> call to my mobile phone and when I answer it - it get disconnected/hangup
> right away. ****
>
>  ****
>
> Why is that? What is the solution to stop that?****
>
>  ****
>
> For example:****
>
>  ****
>
> ACTION: Originate****
>
> Channel: SIP/447XXXXXXX at vpsprovider****
>
> Exten: 210****
>
> Priority: 1****
>
> CallerID: 0044123456789****
>
> Timeout: 60000****
>
> Context: test****
>
>  ****
>
>  ****
>
> exten => 210,1,Answer ****
>
> exten =>
> 210,n,Set(MONITOR_FILENAME=Record-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})***
> *
>
> exten => 210,n,SendDTMF(wwww2w3w)****
>
> exten => 210,n,Monitor(wav,${MONITOR_FILENAME},ib)****
>
> exten => 210,n,Hangup()****
>
>  ****
>
>  ****
>
> Before I had Dial() in the dialplan and it work great and no hangup.  Now
> I am using AMI method.****
>
>  ****
>
>  ****
>
> Thanks****
>
>
> --
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> ** **
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