HI Eric, Nikhil, Thanks a lot for the responses. Bear with me a little as I'm very new to asterisk.
I reproduced the problem using standard dialplan. The following are the configuration files: *chan_dahdi.conf* *[trunkgroups] [channels] language=en nationalprefix=+91 pridialplan=national ; or national or local? usecallerid=yes hidecallerid=no callwaiting=yes allow_call_waiting_calls=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;Might have to play with this. callerid=asreceived facilityenable=yes priindication=outofband cidsignalling=dtmf ; most likely dtmf based on the India link below cidstart=polarity_IN #include dahdi-channels.conf* *extensions.conf:* *[frompstn] exten => xxxxx,1,Ringing exten => xxxxx,n,Dial(Dahdi/G11/yyyyy) exten => xxxxx,n,Hangup()* When aaaaa calls xxxxx, we dial yyyyy This is what I find in the logs: -- Accepting call from 'aaaaa' to 'xxxxx on channel 0/24, span 1 -- Executing [xxxxx@frompstn:1] Ringing("DAHDI/i1/aaaaa-136", "") in new stack -- Executing [xxxxx@frompstn:2] Dial("DAHDI/i1/aaaaaa-136", "Dahdi/G11/yyyyyy") in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/yyyyyy -- DAHDI/i1/yyyyy-137 is proceeding passing it to DAHDI/i1/aaaaa-136 -- DAHDI/i1/yyyyy-137 is ringing # At this point, yyyyy rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/yyyyy-137 is making progress passing it to DAHDI/i1/aaaaa-136 -- Nobody picked up in 30000 ms I'll try out "pri intense debug" during night time when the traffic on our servers is low, and update the list with the logs. In the mean time, is there anything missing in the configuration that rejected calls aren't detected? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 1:10 AM, Eric Wieling <ewiel...@nyigc.com> wrote: > 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 >
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