From: Joshua Colp <jcolp@digium.com>
Date: Wed, 29 Aug 2007 14:13:55 +0000 (+0000)
Subject: (closes issue #9690)
X-Git-Tag: 1.4.12~168
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=8fcc537329d14009bccac1b933feab919b537af9;p=thirdparty%2Fasterisk.git

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 54dc38262b..6fe82e0dca 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -15318,15 +15318,12 @@ restartsearch:
 								ast_mutex_lock(&sip->lock);
 							}
 							if (sip->owner) {
-								if (!(ast_rtp_get_bridged(sip->rtp))) {
-									ast_log(LOG_NOTICE,
-										"Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
-										sip->owner->name,
-										(long) (t - sip->lastrtprx));
-									/* Issue a softhangup */
-									ast_softhangup_nolock(sip->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", sip->owner->name, (long) (t - sip->lastrtprx));
+								ast_log(LOG_NOTICE,
+									"Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
+									sip->owner->name,
+									(long) (t - sip->lastrtprx));
+								/* Issue a softhangup */
+								ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
 								ast_channel_unlock(sip->owner);
 								/* forget the timeouts for this call, since a hangup
 								   has already been requested and we don't want to