Good day.
I have using callback files scheme for bridging 2 numbers thru asterisk with next settings:
callfile:
Channel: Local/1617@dialout-murdozcb
Set: CALLERIDNUM=1617
Context: callback-murdozcb
Extension: 99312
WaitTime: 60
Priority: 1
extesions.conf
[dialout-murdozcb]
exten => h,1,Hangup()
exten => _XXXXXX.,1,Dial(SIP/${EXTEN}@voicetrading)
exten => _XXXXXX.,2,Hangup
[callback-murdozcb]
exten => h,1,Hangup()
exten => _XXXXXX.,1,Answer
exten => _XXXXXX.,2,Playback(beep)
exten => _XXXXXX.,3,Dial(IAX2/1stggateway/${EXTEN})
exten => _XXXXXX.,4,Hangup
[color=blue]Calling process is ok for both sides,
but when 1st call-leg established, script is executing hangup[/color]
but call is not hanging phisically, after this asterisk normally calling to 2nd side and establish bridging.
But i not need this hangup, it is cancelling billing process for 1st leg.
When i am erase 1,Answer directive from 2nd extension, hangup is not doing, but no sound on both side in established call.
What is solution for this problem ?
[color=red]Screen from debug:[/color]
– Attempting call on Local/1617@dialout-murdozcb for 1617@callback-murdozcb:1 (Retry 1)
– Executing [1617@dialout-murdozcb:1] Dial(“Local/1617@dialou
t-murdozcb-407b,2”, “SIP/1617@voicetpovider”) in new stack
– Called 1617@voicetprovider
– Call on SIP/voiceprovider-08cddb20 left from hold
– SIP/voicetprovider-08cddb20 is making progress passing it to Local/1617@dialout-murdozcb-407b,2
– Call on SIP/voiceprovider-08cddb20 left from hold
– SIP/voiceprovider-08cddb20 answered Local/1617@dialout-murdozcb-407
b,2
– Executing [99312@callback-murdozcb:1] Answer(“Local/1617@dia
lout-murdozcb-407b,1”, “”) in new stack
– Executing [99312@callback-murdozcb:2] Playback(“Local/1617@d
ialout-murdozcb-407b,1”, “beep”) in new stack
– <Local/1617@dialout-murdozcb-407b,1> Playing ‘beep’ (language ‘en’
)
== Spawn extension (dialout-murdozcb, 1617, 1) exited non-zero on ‘Local/1617@dialout-murdozcb-407b,2’
– Executing [h@dialout-murdozcb:1] Hangup("Local/1617@dialout-murdoz