|Summary:||ASTERISK-16467: [patch] SIP channel AMI session timeout events feature|
|Reporter:||Kirill Katsnelson (kkm)||Labels:|
|Date Opened:||2010-07-29 14:45:47||Date Closed:||2012-01-20 15:27:01.000-0600|
|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.|
`source` can be either RTPTimeout or SIPSessionTimer.
****** ADDITIONAL INFORMATION ******
There was a little discussion on asterisk-dev that quite died out inconclusively.
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.