]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
authorOlle Johansson <oej@edvina.net>
Wed, 16 May 2007 10:05:47 +0000 (10:05 +0000)
committerOlle Johansson <oej@edvina.net>
Wed, 16 May 2007 10:05:47 +0000 (10:05 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 788a16587a6da84898f8091981dcbe3817ee7d9f..883aaeebbac84ee3273e09377397442d9d44b215 100644 (file)
@@ -17195,7 +17195,7 @@ 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);
        return 0;
 }