[asterisk-dev] Originated Calls and Progress Detection

[Richard Lyman]

Whit Thiele wrote:
>
> 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 )
>
> From what I see, it looks like dsp could be implemented somewhere in 
> the __ast_request_and_dial method, but I’m not sure if this is the 
> best place. Ideas anyone?
>
*snipped

using Async: true in your originate, you can do have a

exten => failed,1,Hangup

in the context for your originate.

it will then generate an

Event: OriginateResponse

that will contain a reason code (not pri causes, just base channel causes).

hope this helps.