[asterisk-commits] jpeeler: branch 1.6.0 r145255 - in /branches/1.6.0: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Sep 30 17:25:43 CDT 2008
Author: jpeeler
Date: Tue Sep 30 17:25:42 2008
New Revision: 145255
URL: http://svn.digium.com/view/asterisk?view=rev&rev=145255
Log:
Merged revisions 145249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r145249 | jpeeler | 2008-09-30 17:21:19 -0500 (Tue, 30 Sep 2008) | 6 lines
(closes issue #13337)
Reported by: pj
Tested by: pj
Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN.
........
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/channels/chan_sip.c
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=145255&r1=145254&r2=145255
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Sep 30 17:25:42 2008
@@ -19647,6 +19647,8 @@
ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
transport = SIP_TRANSPORT_UDP;
}
+ } else { /* use default */
+ transport = SIP_TRANSPORT_UDP;
}
if (!host) {
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