[asterisk-commits] kmoore: trunk r351939 - in /trunk: UPGRADE.txt channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: kmoore
Date: Fri Jan 20 15:26:50 2012
New Revision: 351939

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=351939
Log:
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

Modified:
    trunk/UPGRADE.txt
    trunk/channels/chan_sip.c

Modified: trunk/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE.txt?view=diff&rev=351939&r1=351938&r2=351939
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Fri Jan 20 15:26:50 2012
@@ -52,6 +52,9 @@
 ===
  - 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

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=351939&r1=351938&r2=351939
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jan 20 15:26:50 2012
@@ -26395,6 +26395,8 @@
 				}
 				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 @@
 				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);