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authorMark Michelson <mmichelson@digium.com>2009-04-17 20:20:23 +0000
committerMark Michelson <mmichelson@digium.com>2009-04-17 20:20:23 +0000
commit4bf5e1b805a3cf4ba04beb19fec8f5ced027ee4d (patch)
tree6b78892bb9b64316879ef09e8f24c7872ab2e94e
parent7a7f17ce4ddfe5699986af3534a768ca9773ca02 (diff)
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
-rw-r--r--channels/chan_sip.c6
1 files changed, 1 insertions, 5 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d1b5e946..39b90a901 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;