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| author | jpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-09-30 22:26:51 +0000 |
|---|---|---|
| committer | jpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-09-30 22:26:51 +0000 |
| commit | 74ffefa22e4e83a763fa5245edabce161f1f0872 (patch) | |
| tree | c605357b5bec179be156fffbbb4f10c6bd69e61b | |
| parent | eb819fa634d6b34f6df0902cc778a60605d1505b (diff) | |
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.
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@145262 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 5bebd1d78..e16370e56 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -20263,6 +20263,8 @@ 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) { |