[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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