[asterisk-users] Capturing call Reject/Decline events on a PRI line

[DHAVAL INDRODIYA]

Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,

also read this for better implementation.

http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause

regards
Dhaval

On Fri, Jul 29, 2011 at 11:58 AM, Nikhil <d.nikhil at cem-solutions.net> wrote:

> **
> find the inline comment...
>
>
> On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
>
> The dialplan is very simple. When the call comes in, we hand the call over
> to adhearsion.
> This is how the dialplan looks:
>
> ;group 0 will be used for incoming calls
> EXOIN = DAHDI/g0
>
> ;group 11 for outgoing
> EXOOUT = DAHDI/G11
>
> ;This will be used by adhearsion
> EXOCID=xxxxxxxx
>
> [general]
> autofallthrough = yes ;really?
> clearglobalvars = no
>
> [frompstn]
> ;Send everything to adhearsion
> exten => _X.,1,Ringing
> exten => _X.,n,AGI(agi://127.0.0.1)
>
>     exten => _X.,n,Hangup() ; Please try this.
>
>
> ; End dialplan
>
> The rest of the logic happens in adhearsion.
>
> --
> Thanks,
> Ishwar.
>
>
> On Thu, Jul 28, 2011 at 6:33 PM, Nikhil <d.nikhil at cem-solutions.net>wrote:
>
>>  Can you share the dialplan ,where SIP call is dialing...
>> Thanks
>> Nikhil
>>
>>
>> On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
>>
>>  Hello everybody,
>>
>> We have an asterisk 1.8.4.1 setup, connected to a PRI line.
>>
>> We're currently facing an issue where asterisk does not recognise the
>> event when the called party declines/cuts the call. This happens
>> specifically over calls on a PRI line. For calls over SIP, call decline
>> event is captured properly.
>>
>> I wasn't able to find a solution on the asterisk-users mailing list
>> archive. Any suggestions/help would be much appreiciated :) I can share the
>> relevant parts of the configuration files, if needed.
>>
>> Here's an excerpt from asterisk logs for a SIP call.
>>     -- SIP/xxxxx-00000000 requested special control 16, passing it to
>> SIP/xxxxx-00000001
>>     -- Started music on hold, class 'default', on SIP/xxxxx-00000001
>>     -- SIP/xxxxx-00000000 requested special control 20, passing it to
>> SIP/xxxxx-00000001
>>     -- Got SIP response 603 "Decline" back from 127.0.0.1:5063
>>     -- SIP/xxxxx-00000001 is busy
>>     -- Stopped music on hold on SIP/xxxxx-00000001
>>
>> As you can see, on a SIP call, a call reject event is identified.
>>
>> For a call over the PRI, on the other hand, this event is not recognised.
>> Here's an excerpt from asterisk log for a call over PRI.
>> Call from yyyy to xxxx.
>>     -- Requested transfer capability: 0x10 - 3K1AUDIO
>>     -- Called G11/xxxxx
>>     -- Started music on hold, class 'default', on DAHDI/i1/yyyyy
>>     -- DAHDI/i1/xxxxx-18f8 is proceeding passing it to DAHDI/i1/yyyyy
>>     -- DAHDI/i1/xxxxx-18f8 is ringing
>> # At this point in time, xxxxx rejects the call. The event that's logged
>> in asterisk is the following:
>>     -- DAHDI/i1/xxxxx-18f8 is making progress passing it to DAHDI/i1/yyyyy
>> # And the call times out after the default 30s.
>>     -- Nobody picked up in 30000 ms
>>
>> Is there a reason why asterisk doesn't recognise the "call decline", and
>> does it need any configuration changes to enable this?
>>
>> Thanks for your help.
>>
>> --
>> Cheers,
>> Ishwar.
>>
>>
>> --
>> _____________________________________________________________________
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>>
>>
>> --
>> _____________________________________________________________________
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>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
> --
> _____________________________________________________________________
> -- 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
>
>
>
> --
> _____________________________________________________________________
> -- 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
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