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

[Ishwar Sridharan]

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)

; 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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