1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk DAHDI configs. 2) Can't troubleshoot when everything important is masked by an AGI script. Reproduce the problem using standard dialplan stuff.
> -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Ishwar Sridharan > Sent: Thursday, July 28, 2011 2:52 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI > line > > Hi Eric, > > There weren't any lines with "PRI channel =>" in the chan_dahdi.conf > > However, I added the lines you'd mentioned, near the top of the file. Still, > no difference in either the behaviour or the asterisk output. > > Please note that as soon as the call lands on asterisk, we pass the control > over to adhearsion. Does that affect how events are handled in asterisk? > > -- > Thanks, > Ishwar. > > > > On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling <ewiel...@nyigc.com> wrote: > > > > > > -----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 > > -- _____________________________________________________________________ -- 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
