[asterisk-users] Asterisk CallCompletion dialplan

satish patel satish_lx at hotmail.com
Tue Feb 8 11:01:43 CST 2011


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 
 		 	   		  
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