[asterisk-dev] Originated Calls and Progress Detection

[Whit Thiele]

I've seen callprogress in zapata.conf , but it seems to be used/applicable
with analog Zap channels. What I am hoping to do is accomplish the same kind
of feature on PRI and SIP lines where there the channel is never
"Connected". The originated call fails but I'm trying to figure out how to
determine why. For instance, how do you determine what was a No Answer (the
call rings until it times out) versus a Disconnected Number (the other side
plays a "Number not in service" message) 

 

The OriginateResponse Event is not giving the correct 'reason' code (Reason
is always 0 )