> -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Nikhil > Sent: Thursday, July 28, 2011 9:03 AM > To: asterisk-users@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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
