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/1.6.2@189105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -19426,11 +19426,7 @@ 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! */
 		const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");