[asterisk-commits] russell: branch group/sip-tcptls r99075 - /team/group/sip-tcptls/channels/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 18 12:46:35 CST 2008
Author: russell
Date: Fri Jan 18 12:46:35 2008
New Revision: 99075
URL: http://svn.digium.com/view/asterisk?view=rev&rev=99075
Log:
Fix a few things that got out of sync with trunk
Modified:
team/group/sip-tcptls/channels/chan_sip.c
Modified: team/group/sip-tcptls/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/sip-tcptls/channels/chan_sip.c?view=diff&rev=99075&r1=99074&r2=99075
==============================================================================
--- team/group/sip-tcptls/channels/chan_sip.c (original)
+++ team/group/sip-tcptls/channels/chan_sip.c Fri Jan 18 12:46:35 2008
@@ -16572,13 +16572,15 @@
sip_pvt_unlock(targetcall_pvt);
if (res) {
/* Failed transfer */
- /* Could find better message, but they will get the point */
- transmit_notify_with_sipfrag(transferer, seqno, "486 Busy", TRUE);
+ transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
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 */
- ast_hangup(transferer->owner);
+ if (res != -2)
+ ast_hangup(transferer->owner);
+ else
+ ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
} else {
/* Transfer succeeded! */
@@ -16874,7 +16876,6 @@
be accessible after the transfer! */
*nounlock = 1;
ast_channel_unlock(current.chan1);
- ast_channel_unlock(current.chan2);
/* Connect the call */
@@ -18016,6 +18017,10 @@
return;
/* If the call is not in UP state or redirected outside Asterisk, no need to check timers */
if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
+ return;
+
+ /* If the call is involved in a T38 fax session do not check RTP timeout */
+ if (dialog->t38.state == T38_ENABLED)
return;
/* If we have no timers set, return now */
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