Hi All,
I’m trying to do “things” after my Dial application terminates (e.g. play IVR
to called party, calling party, etc.). I’m trying to use the local channel for
this purpose but so far with no success. I’m using 1.6.1.18 and this is my
extensions.conf:
[Internal]
exten => _22,1,Dial(Local/${ext...@cw/n) ; 22 is test number
exten => _22,2,Noop(After Hangup)
[CW]
exten => _x.,1,Dial(SIP/307)
exten => _x.,2,Noop(After Hangup)
The call never reaches neither of the Noop applications. Consol:
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [...@internal:1] Dial("SIP/309-000000a5", "Local/2...@cw/n")
in new stack
-- Called 2...@cw/n
-- Executing [...@cw:1] Dial("Local/2...@cw-af6f;2", "SIP/307") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called 307
-- SIP/307-000000a6 is ringing
-- Local/2...@cw-af6f;1 is ringing
-- SIP/307-000000a6 is ringing
-- SIP/307-000000a6 is ringing
-- SIP/307-000000a6 is ringing
-- SIP/307-000000a6 answered Local/2...@cw-af6f;2
-- Local/2...@cw-af6f;1 answered SIP/309-000000a5
== Spawn extension (CW, 22, 1) exited non-zero on 'Local/2...@cw-af6f;2'
== Spawn extension (Internal, 22, 1) exited non-zero on 'SIP/309-000000a5'
If I use the ‘g’ option in my Dial() both Noop will be run only if the called
party hang-up first. I need a simple continuation in the dial plan regardless
of who disconnected the call.
Thanks in advance
Harel
--
_____________________________________________________________________
-- 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