[asterisk-dev] Originated Calls and Progress Detection

Whit Thiele dev at whit.ca
Sun Dec 9 18:03:56 CST 2007


Richard,

Thanks for the suggestion. The OriginateResponse event is generated
automatically. 

I've also used the failed extension and tried to look at both DIALSTATUS and
HANGUPCAUSE variables, but they don't appear to get set properly either.
There was a similar thread on this strategy back in February, 2007 which had
a couple other dialplan suggestions (like looking at DIALSTATUS using the
'h' extension) None of these suggestions yielded any solutions.

On my test system, HANGUPCAUSE is always set to 16 and the OriginateResponse
reason is 0 for both a real No Answer and a Disconnected Number. Even the
CDR records show NO ANSWER for both calls. 

This is why I'm trying to look into changing the code itself (hence the
posting on this list) for a solution. 

So the question seems to be where the best place is to attach a dsp process
to determine what's going on during the call launch.. unless there is
something else to try that I've missed.







-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Richard Lyman
Sent: Sunday, December 09, 2007 10:18 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Originated Calls and Progress Detection

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.



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