Files
asterisk/channels
Kinsey Moore c6fd4f5d74 SIP session timeout AMI event
Add 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 description:

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

`source` can be either RTPTimeout or SIPSessionTimer

(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-20 21:26:50 +00:00
..
2009-08-10 19:20:57 +00:00
2012-01-14 15:51:43 +00:00
2012-01-20 21:26:50 +00:00
2010-06-07 20:04:42 +00:00