diff options
| author | Kinsey Moore <kmoore@digium.com> | 2012-01-20 21:26:50 +0000 |
|---|---|---|
| committer | Kinsey Moore <kmoore@digium.com> | 2012-01-20 21:26:50 +0000 |
| commit | c6fd4f5d7401633649bbf2b45f57f3ddc6ae18f1 (patch) | |
| tree | 974e01b8dd5f1c0a10396b9df1cdb815738be667 /UPGRADE.txt | |
| parent | 778fa4abafbb19eeb7690c51d886908c0040cf9c (diff) | |
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
Diffstat (limited to 'UPGRADE.txt')
| -rw-r--r-- | UPGRADE.txt | 3 |
1 files changed, 3 insertions, 0 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index e5a81a98d..83c2b2b6c 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -52,6 +52,9 @@ SIP === - A new option "tonezone" for setting default tonezone for the channel driver or individual devices + - A new manager event, "SessionTimeout" has been added and is triggered when + a call is terminated due to RTP stream inactivity or SIP session timer + expiration. users.conf: - A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten |