Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,also read this for better implementation. http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause regards Dhaval On Fri, Jul 29, 2011 at 11:58 AM, Nikhil <[email protected]> wrote: > ** > find the inline comment... > > > On 07/29/2011 12:11 AM, Ishwar Sridharan wrote: > > 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) > > exten => _X.,n,Hangup() ; Please try this. > > > ; End dialplan > > The rest of the logic happens in adhearsion. > > -- > Thanks, > Ishwar. > > > On Thu, Jul 28, 2011 at 6:33 PM, Nikhil <[email protected]>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 >> > > > -- > _____________________________________________________________________ > -- 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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