From c70c0f1df070a862648e2f26e3e5a85ac43dcf9d Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Fri, 17 Apr 2009 20:21:42 +0000 Subject: [PATCH] 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 | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index becd3f8545..e88cae3fa5 100644 --- a/channels/chan_sip.c +++ b/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");