[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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