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| author | Olle Johansson <oej@edvina.net> | 2007-05-16 10:09:42 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2007-05-16 10:09:42 +0000 |
| commit | c472b899efe2bb8e1c7a2a770071ee0b455cefdf (patch) | |
| tree | 7c577b9af28a2b8f6eb9ad51ecc81b0d0c39624e | |
| parent | 69d15e3451b29c3ee7dec46ea8a270366311cf75 (diff) | |
Merged revisions 64578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | channels/chan_sip.c | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 56d5d176b..380f711eb 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -18151,9 +18151,9 @@ static int sip_sipredirect(struct sip_pvt *p, const char *dest) transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq); sip_scheddestroy(p, 32000); /* Make sure we stop send this reply. */ - + sip_alreadygone(p); /* hangup here */ - return -1; + return 0; } /*! \brief Return SIP UA's codec (part of the RTP interface) */ |