diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-04-17 20:21:05 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-04-17 20:21:05 +0000 |
| commit | 824f253939810d288069562a05f7eb3cc0ffe8f7 (patch) | |
| tree | 9335721bc82f6f721deaf404065f71f796ff84c6 /channels | |
| parent | 25bf87abbf56a4d197433e4dbe8b421ad18f2b28 (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/1.6.0@189101 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -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 a7c209955..f0e3547e6 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -17658,11 +17658,7 @@ 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! */ |