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

[Nikhil]

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 <http://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 
> <mailto: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
>>     <http://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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>
>
>
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