Hi Users,
I'm planing to implement call completion feature in asterisk 1.8 but having
some issue. I am following this document
https://wiki.asterisk.org/wiki/display/AST/Generic+Call+Completion+Example
I am getting error non-zero error on console. I am using softphone x-lite
root at tux:/etc/asterisk# asterisk -r
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [30 at from-sip:1]
CallCompletionRequest("SIP/7623-00000013", "") in new stack
== Spawn extension (from-sip, 30, 1) exited non-zero on
'SIP/7623-00000013'
sip.conf
[Mark]context=phone_callscc_agent_policy=genericcc_monitor_policy=generic ;We
will accept defaults for the rest of the cc parameters;We also are not concerned
with other SIP details for this;example
[Richard]context=phone_callscc_agent_policy=genericcc_monitor_policy=generic
extensions.conf
[phone_calls]exten => 1000,1,Dial(SIP/Mark,20)exten => 1000,n,Hangupexten
=> 2000,1,Dial(SIP/Richard,20)exten => 2000,n,Hangupexten =>
30,1,CallCompletionRequestexten => 30,n,Hangupexten =>
31,1,CallCompletionCancelexten => 31,n,Hangup
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110208/db6f6145/attachment.htm>
