asterisk users - Jul 2011 - Capturing call Reject/Decline events on a PRI line

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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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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On Thursday 28 Jul 2011, Ishwar Sridharan wrote:
> Is there a reason why asterisk doesn't recognise the "call
decline", and
> does it need any configuration changes to enable this?
What are you seeing for ${HANGUPCAUSE} when this happens ?  (Put a line such 
as

exten => yyyyy, n, NoOp(Hangup cause was ${HANGUPCAUSE})

in your extensions.conf after the Dial() statement.)

Note that with traditional phones, there is actually no way to decline a call 
besides answering it and asking the caller politely to go away  :)

-- 
AJS

Answers come *after* questions.

> There is no event for Asterisk to recognize. The PROGRESS message just
> says that there is an audio message available for the caller to listen
> to. Asterisk just passes the indication to the peer channel and opens
> the audio path. It is the caller who must recognize any audio message
> that their call has been dropped.
> 
> Thanks for the explanation. Any suggestion on how to recognise that
> the call has been dropped?
> 
> 
> As far as ISDN is concerned, the
> call has not been answered yet so Asterisk must keep waiting.
> 
As far as the ISDN signaling is concerned, the call is still going.
There is no signaling to indicate the call is not going to proceed
any further.

Richard