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

[Eric Wieling]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Nikhil
> Sent: Thursday, July 28, 2011 9:03 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI
> line
> 
> 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.


Try adding the following before your PRI channel => lines in your chan_dahdi.conf.  If you are using a GUI like FreePBX, you will have place the info where you need to for FreePBX.

facilityenable=yes
priindication=outofband