diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-04-17 20:21:42 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-04-17 20:21:42 +0000 |
| commit | 6f99db2272a85df88f2268f69379b6de75988be1 (patch) | |
| tree | 1d84de271ed2bb124bec473102799aa2681d2709 | |
| parent | 2774a34d977afb1ff61d94b2a491169e0ea6fd67 (diff) | |
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@189105 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 6 |
1 files changed, 1 insertions, 5 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index becd3f854..e88cae3fa 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"); |