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 7a325a17c866ad2555b8b28fb4668196123f8f8c..62ab783773e0a6814f4839f17d7f9d62c1903676 100644 (file)
--- 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! */