Re-enable RTCP quality reports

(Bug found in SIP Master Class, Chicago)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37474 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -3150,6 +3150,7 @@ static int sip_hangup(struct ast_channel *ast)
 	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 @@ static int sip_hangup(struct ast_channel *ast)
 					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. */