Merged revisions 81331 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4 lines

(closes issue #9690)
Reported by: mattv
Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81332 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -16427,13 +16427,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
 					usleep(1);
 					sip_pvt_lock(dialog);
 				}
-				if (!(ast_rtp_get_bridged(dialog->rtp))) {
-					ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
-						dialog->owner->name, (long) (t - dialog->lastrtprx));
-					/* Issue a softhangup */
-					ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
-				} else
-					ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", dialog->owner->name, (long) (t - dialog->lastrtprx));
+				ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
+					dialog->owner->name, (long) (t - dialog->lastrtprx));
+				/* Issue a softhangup */
+				ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
 				ast_channel_unlock(dialog->owner);
 				/* forget the timeouts for this call, since a hangup
 				   has already been requested and we don't want to