From 6281f896133f501833cb381ce1ffc0fd7eb3f29e Mon Sep 17 00:00:00 2001
From: jpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b>
Date: Tue, 30 Sep 2008 22:25:42 +0000
Subject: 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.

........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@145255 f38db490-d61c-443f-a65b-d21fe96a405b
---
 channels/chan_sip.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'channels')

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6440c23ef..ae3f7b3aa 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -19647,6 +19647,8 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
 				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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