From c6fd4f5d7401633649bbf2b45f57f3ddc6ae18f1 Mon Sep 17 00:00:00 2001 From: Kinsey Moore Date: Fri, 20 Jan 2012 21:26:50 +0000 Subject: 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 --- UPGRADE.txt | 3 +++ 1 file changed, 3 insertions(+) (limited to 'UPGRADE.txt') 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 -- cgit v1.2.3