From: Mark Michelson <mmichelson@digium.com>
Date: Fri, 17 Apr 2009 20:21:26 +0000 (+0000)
Subject: Merged revisions 189097 via svnmerge from
X-Git-Tag: 1.6.1.0-rc5~8
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=6af217578ee83c977f2fe68018194b3a0ad1353f;p=thirdparty%2Fasterisk.git

Merged revisions 189097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines

  Prevent a crash when SIP blonde transferring an unbridged call.

  If one attempts to use the attended transfer button on a SIP phone
  to transfer an unbridged call (such as a call to an IVR) but hangs
  up while the target of the transfer is still ringing, we need to not
  crash.

  The problem was that ast_hangup was called from outside the channel
  thread.

  AST-211
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7a325a17c8..62ab783773 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -18529,11 +18529,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
 		append_history(transferer, "Xfer", "Refer failed");
 		if (targetcall_pvt->owner)
 			ast_channel_unlock(targetcall_pvt->owner);
-		/* Right now, we have to hangup, sorry. Bridge is destroyed */
-		if (res != -2)
-			ast_hangup(transferer->owner);
-		else
-			ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
+		ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
 	} else {
 		/* Transfer succeeded! */