Prevent a crash when SIP blonde transferring an unbridged call.
authorMark Michelson <mmichelson@digium.com>
Fri, 17 Apr 2009 20:20:23 +0000 (20:20 +0000)
committerMark Michelson <mmichelson@digium.com>
Fri, 17 Apr 2009 20:20:23 +0000 (20:20 +0000)
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/trunk@189097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 5d1b5e9..39b90a9 100644 (file)
@@ -20067,11 +20067,7 @@ 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 {
                struct ast_party_connected_line connected_caller;