[asterisk-commits] trunk r37474 - /trunk/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Wed Jul 12 14:09:55 2006
New Revision: 37474

URL: http://svn.digium.com/view/asterisk?rev=37474&view=rev
Log:
Re-enable RTCP quality reports
(Bug found in SIP Master Class, Chicago)

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=37474&r1=37473&r2=37474&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Jul 12 14:09:55 2006
@@ -3150,6 +3150,7 @@
 	struct sip_pvt *p = ast->tech_pvt;
 	int needcancel = FALSE;
 	struct ast_flags locflags = {0};
+	struct ast_channel oldowner = ast;
 
 	if (!p) {
 		ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n");
@@ -3260,10 +3261,10 @@
 					if (p->vrtp)
 						append_history(p, "RTCPvideo", "Quality:%s", videoqos);
 				}
-				if (p->rtp && p->owner)
-					pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
-				if (p->vrtp && p->owner)
-					pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
+				if (p->rtp && oldowner)
+					pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
+				if (p->vrtp && oldowner)
+					pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
 			} else {
 				/* Note we will need a BYE when this all settles out
 				   but we can't send one while we have "INVITE" outstanding. */