[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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