[asterisk-dev] Detailed SIP disconnect reason

[Kirill 'Big K' Katsnelson]

On 100222 2228, Olle E. Johansson wrote:
> 23 feb 2010 kl. 00.08 skrev CDR:
> 
>> I support to export the disconect reason in a variable, in numeric form, so we may send it back to the caller. Right now, if the second leg returns a "bad number" or any other message, we cannot convey such information to the caller.
> Isn't this feature already merged into trunk?

Are you talking about [226687] that both transmits and decodes Reason: 
Q.850;cause... header?

The feature that I am missing is a little different. In both 
proc_session_timer() and check_rtp_timeout(), if timeout expires, soft 
hangup is scheduled, without any trace as to why. I looked up the 
functions in the latest trunk, and they have not principally changed 
from those in 1.6.

In the end, the call is disconnected with 16 "normal teardown" message, 
even if killed by the RTP timer.

I want to be able to access the "deeper" hangup reason: whether a call 
was disconnected by a timeout (RTP, session timer), or, if terminated 
normally, then which side of the call sent the BYE. Right now, I cannot 
see any difference between these events in the dialplan.

Am I missing anything?

  -kkm
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