[asterisk-commits] file: branch 1.4 r81331 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Wed Aug 29 09:13:55 2007
New Revision: 81331

URL: http://svn.digium.com/view/asterisk?view=rev&rev=81331
Log:
(closes issue #9690)
Reported by: mattv
Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack.

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=81331&r1=81330&r2=81331
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Aug 29 09:13:55 2007
@@ -15318,15 +15318,12 @@
 								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