I have a problem with call groups..
I'm running asterisk 1.6 and sail 3.1.0.97 ( and was the same on .93)
For my Trunklines I send the incoming call to a ring group, defined in call groups including a bunch of extensions.
If I ring the call group from an extension, it works..
== Using SIP RTP CoS mark 5
-- Executing [55@internal:1] AGI("SIP/5021-00000004", "sarkhpe,Alias,SIP/5021 SIP/5020 SIP/5010 SIP/5029,55") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Dial) Options: (SIP/5021&SIP/5020&SIP/5010&SIP/5029,15,)
== Using SIP RTP CoS mark 5
-- Called 5021
== Using SIP RTP CoS mark 5
[Jan 5 13:47:03] WARNING[4495]: app_dial.c:1750 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[Jan 5 13:47:03] WARNING[4495]: app_dial.c:1750 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
-- Called 5029
-- SIP/5029-00000006 is ringing
-- SIP/5021-00000005 is ringing
-- AGI Script Executing Application: (Answer) Options: ()
-- <SIP/5021-00000004>AGI Script sarkhpe completed, returning 0
-- Executing [h@extensions:1] Hangup("SIP/5021-00000004", "") in new stack
== Spawn extension (extensions, h, 1) exited non-zero on 'SIP/5021-00000004'
but when the Trunkline rings.. the caller hears engaged tone:-
== Using SIP RTP CoS mark 5
-- Executing [61312345551@mainmenu:1] AGI("SIP/peer3447-00000007", "sarkhpe,Inbound") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(userfield)=61312345551)
-- <SIP/peer3447-00000007>AGI Script sarkhpe completed, returning 0
-- Auto fallthrough, channel 'SIP/peer3447-00000007' status is 'UNKNOWN'
-- Executing [h@mainmenu:1] Hangup("SIP/peer3447-00000007", "") in new stack
== Spawn extension (mainmenu, h, 1) exited non-zero on 'SIP/peer3447-00000007'
Yet if I target a single extension as the trunkline outcome, it works ! ....
== Using SIP RTP CoS mark 5
-- Executing [61312345551@mainmenu:1] AGI("SIP/peer3447-00000008", "sarkhpe,Inbound") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(userfield)=61312345551)
-- AGI Script Executing Application: (Answer) Options: ()
-- <SIP/peer3447-00000008>AGI Script sarkhpe completed, returning 0
-- Executing [5021@extensions:1] AGI("SIP/peer3447-00000008", "sarkhpe,InCall,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Dial) Options: (SIP/5021,20,kt)
== Using SIP RTP CoS mark 5
-- Called 5021
-- SIP/5021-00000009 is ringing
Curioustly if my MNF trunk rings in. I get...
== Using SIP RTP CoS mark 5
[Jan 5 13:57:37] NOTICE[3867]: chan_sip.c:20291 handle_request_invite: Call from '09109123' to extension '09109123' rejected because extension not found in context 'mainmenu'.
== Using SIP RTP CoS mark 5
[Jan 5 13:57:38] NOTICE[3867]: chan_sip.c:20291 handle_request_invite: Call from '09109123' to extension '09109123' rejected because extension not found in context 'mainmenu'.
So for some reason this Trunkline wants to route the call to its own 'peer' instead of the defined extension or trunkline.