Summary:ASTERISK-16467: [patch] SIP channel AMI session timeout events feature
Reporter:Kirill Katsnelson (kkm)Labels:
Date Opened:2010-07-29 14:45:47Date Closed:2012-01-20 15:27:01.000-0600
Versions:Frequency of
Environment:Attachments:( 0) 017754-chansip-sessiontimeoutevents-trunk.diff
Description:This simple patch adds an AMI event in the Call category, that is issued when a call is terminated due to either RTP stream inactivity or SIP session timer expiration.

Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id

`source` can be either RTPTimeout or SIPSessionTimer.


There was a little discussion on asterisk-dev that quite died out inconclusively.

See http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg41860.html

In summary, Mark Michelson sent a "giant +1 on all counts", but Olle Johansson thought this should rather be part of a bigger thing of hangup cause reporting. My own opinion is that unless the bigger thing is making it into 1.8, then this lesser feature should.
Comments:By: Paul Belanger (pabelanger) 2010-07-29 20:20:48

We'll need some documentation

By: Kirill Katsnelson (kkm) 2010-07-29 22:19:43

Absolutely. Where should the events be described? There is not much in doc/text/manager.tex, and I cannot find relevant docs anywhere else.

By: Leif Madsen (lmadsen) 2010-08-05 14:19:25

I agree that larger scope of including this in hangup reporting would be ideal, but this seems like a trivial and useful change for the time being.