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

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) */