From c6fd4f5d7401633649bbf2b45f57f3ddc6ae18f1 Mon Sep 17 00:00:00 2001
From: Kinsey Moore <kmoore@digium.com>
Date: Fri, 20 Jan 2012 21:26:50 +0000
Subject: [PATCH] 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 +++
channels/chan_sip.c | 4 ++++
2 files changed, 7 insertions(+)
diff --git a/UPGRADE.txt b/UPGRADE.txt
index e5a81a98d0..83c2b2b6cb 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
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1ea54538b4..e80d2d913b 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -26395,6 +26395,8 @@ static int check_rtp_timeout(struct sip_pvt *dialog, time_t t)
}
ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
ast_channel_name(dialog->owner), (long) (t - dialog->lastrtprx));
+ manager_event(EVENT_FLAG_CALL, "SessionTimeout", "Source: RTPTimeout\r\n"
+ "Channel: %s\r\nUniqueid: %s\r\n", ast_channel_name(dialog->owner), dialog->owner->uniqueid);
/* Issue a softhangup */
ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(dialog->owner);
@@ -26647,6 +26649,8 @@ static int proc_session_timer(const void *vp)
sip_pvt_lock(p);
}
+ manager_event(EVENT_FLAG_CALL, "SessionTimeout", "Source: SIPSessionTimer\r\n"
+ "Channel: %s\r\nUniqueid: %s\r\n", ast_channel_name(p->owner), p->owner->uniqueid);
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(p->owner);
sip_pvt_unlock(p);
--
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