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

[Ishwar Sridharan]

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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