| 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 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| 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. ****** ADDITIONAL INFORMATION ****** 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. | ||