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

@@ -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

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);