[asterisk-commits] file: trunk r81332 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Aug 29 09:16:08 CDT 2007
Author: file
Date: Wed Aug 29 09:16:07 2007
New Revision: 81332
URL: http://svn.digium.com/view/asterisk?view=rev&rev=81332
Log:
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.
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=81332&r1=81331&r2=81332
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Aug 29 09:16:07 2007
@@ -16427,13 +16427,10 @@
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
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