The dialplan is very simple. When the call comes in, we hand the call over to adhearsion. This is how the dialplan looks:
;group 0 will be used for incoming calls EXOIN = DAHDI/g0 ;group 11 for outgoing EXOOUT = DAHDI/G11 ;This will be used by adhearsion EXOCID=xxxxxxxx [general] autofallthrough = yes ;really? clearglobalvars = no [frompstn] ;Send everything to adhearsion exten => _X.,1,Ringing exten => _X.,n,AGI(agi://127.0.0.1) ; End dialplan The rest of the logic happens in adhearsion. -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:33 PM, Nikhil <d.nik...@cem-solutions.net> wrote: > ** > 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 > -- 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. > > > -- > _____________________________________________________________________ > -- 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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