Hi,
I need to generate 2 calls to an external destination over a SIP trunk, play a message on one trunk while recording on the other. Both channels would need to be hungup at the same time. I am currently using .call files and transfering them to the /var/spool/asterisk/outgoing/ folder in order to generate calls. I have a description of the way i am trying it at the moment below. What i am wondering is am i going about this the right way(would originate be better) and would anyone have a sugestion on how to end the calls at the same time?
Many Thanks for any help.
A simple explination of the way i have it currently is this:
play-audio.call
Context: play
Extension: 100
Priority: 1
play context
[play]
exten => 100,1,Answer
exten => 100,2,Wait(1)
exten => 100,3,Dial(SIP/......)
exten => 100,4,Playback(soundfile)
record.call
Context: record
Extension: 101
Priority: 1
record context
[record]
exten => 100,1,Answer
exten => 100,2,Wait(1)
exten => 100,3,Monitor(wav,myfilename)
exten => 100,4,Dial(SIP/......)