[asterisk-commits] file: trunk r69795 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jun 18 14:02:46 CDT 2007
Author: file
Date: Mon Jun 18 14:02:45 2007
New Revision: 69795
URL: http://svn.digium.com/view/asterisk?view=rev&rev=69795
Log:
Merged revisions 69794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 lines
Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)
........
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=69795&r1=69794&r2=69795
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jun 18 14:02:45 2007
@@ -15850,6 +15850,10 @@
if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
return;
+ /* If the call is involved in a T38 fax session do not check RTP timeout */
+ if (dialog->t38.state == T38_ENABLED)
+ return;
+
/* If we have no timers set, return now */
if (ast_rtp_get_rtpkeepalive(dialog->rtp) == 0 || (ast_rtp_get_rtptimeout(dialog->rtp) == 0 && ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
return;
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