From: Mark Michelson <mmichelson@digium.com>
Date: Fri, 17 Apr 2009 20:20:23 +0000 (+0000)
Subject: Prevent a crash when SIP blonde transferring an unbridged call.
X-Git-Tag: 11.0.0-beta1~5035
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=4bf5e1b805a3cf4ba04beb19fec8f5ced027ee4d;p=thirdparty%2Fasterisk.git

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/trunk@189097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d1b5e946e..39b90a901d 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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;