[svn-commits] twilson: trunk r369559 - in /trunk: ./ channels/ channels/sip/include/

[SVN commits to the Digium repositories]

Author: twilson
Date: Tue Jul  3 09:49:19 2012
New Revision: 369559

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369559
Log:
Better handle re-INVITEs with provisional but no final repsonses

A previous attempt at fixing this issue had negative side effects related
to attended transfers which this patch should resolve. Many thanks to
Steve Davies for all of the good suggestions and testing.

(closes issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies, Terry Wilson
Review: https://reviewboard.asterisk.org/r/2009/
........

Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369558 from http://svn.asterisk.org/svn/asterisk/branches/10

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c
    trunk/channels/sip/include/sip.h

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=369559&r1=369558&r2=369559
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jul  3 09:49:19 2012
@@ -6374,6 +6374,21 @@
 	return 0;
 }
 
+static int reinvite_timeout(const void *data)
+{
+	struct sip_pvt *dialog = (struct sip_pvt *) data;
+	struct ast_channel *owner = sip_pvt_lock_full(dialog);
+	check_pendings(dialog);
+	dialog->reinviteid = -1;
+	if (owner) {
+		ast_channel_unlock(owner);
+		ast_channel_unref(owner);
+	}
+	ao2_unlock(dialog);
+	dialog_unref(dialog, "unref for reinvite timeout");
+	return 0;
+}
+
 /*! \brief  sip_hangup: Hangup SIP call
  * Part of PBX interface, called from ast_hangup */
 static int sip_hangup(struct ast_channel *ast)
@@ -6497,7 +6512,7 @@
 				stop_session_timer(p);
 			}
 
-			if (!p->pendinginvite || p->ongoing_reinvite) {
+			if (!p->pendinginvite) {
 				struct ast_channel *bridge = ast_bridged_channel(oldowner);
 				char quality_buf[AST_MAX_USER_FIELD], *quality;
 
@@ -6559,8 +6574,16 @@
 				ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
 				ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);	
 				AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
-				if (sip_cancel_destroy(p))
+				if (sip_cancel_destroy(p)) {
 					ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+				}
+				/* If we have an ongoing reinvite, there is a chance that we have gotten a provisional
+				 * response, but something weird has happened and we will never receive a final response.
+				 * So, just in case, check for pending actions after a bit of time to trigger the pending
+				 * bye that we are setting above */
+				if (p->ongoing_reinvite && p->reinviteid < 0) {
+					p->reinviteid = ast_sched_add(sched, 32 * p->timer_t1, reinvite_timeout, dialog_ref(p, "ref for reinvite_timeout"));
+				}
 			}
 		}
 	}
@@ -7971,6 +7994,7 @@
 	p->method = intended_method;
 	p->initid = -1;
 	p->waitid = -1;
+	p->reinviteid = -1;
 	p->autokillid = -1;
 	p->request_queue_sched_id = -1;
 	p->provisional_keepalive_sched_id = -1;
@@ -21036,8 +21060,9 @@
 			   INVITE, but do set an autodestruct just in case we never get it. */
 		} else {
 			/* We have a pending outbound invite, don't send something
-				new in-transaction */
-			if (p->pendinginvite)
+			 * new in-transaction, unless it is a pending reinvite, then
+			 * by the time we are called here, we should probably just hang up. */
+			if (p->pendinginvite && !p->ongoing_reinvite)
 				return;
 
 			if (p->owner) {
@@ -21286,12 +21311,19 @@
  		p->invitestate = INV_COMPLETED;
 	}
  	
+	if ((resp >= 200 && reinvite)) {
+		p->ongoing_reinvite = 0;
+		if (p->reinviteid > -1) {
+			AST_SCHED_DEL_UNREF(sched, p->reinviteid, dialog_unref(p, "unref dialog for reinvite timeout because of a final response"));
+			/* Since we got a final response to the reinvite, but were relying on the reinvite_timeout
+			 * function to clean up after the reinvite, we need to make sure and call check_pendings */
+			check_pendings(p);
+		}
+	}
+
 	/* Final response, clear out pending invite */
 	if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
 		p->pendinginvite = 0;
-		if (reinvite) {
-			p->ongoing_reinvite = 0;
-		}
 	}
 
 	/* If this is a response to our initial INVITE, we need to set what we can use

Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=369559&r1=369558&r2=369559
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Tue Jul  3 09:49:19 2012
@@ -1140,6 +1140,7 @@
 
 	int initid;                         /*!< Auto-congest ID if appropriate (scheduler) */
 	int waitid;                         /*!< Wait ID for scheduler after 491 or other delays */
+	int reinviteid;                     /*!< Reinvite in case of provisional, but no final response */
 	int autokillid;                     /*!< Auto-kill ID (scheduler) */
 	int t38id;                          /*!< T.38 Response ID */
 	struct sip_refer *refer;            /*!< REFER: SIP transfer data structure */