[svn-commits] trunk r36128 - /trunk/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon Jun 26 12:07:32 MST 2006
Author: oej
Date: Mon Jun 26 14:07:32 2006
New Revision: 36128
URL: http://svn.digium.com/view/asterisk?rev=36128&view=rev
Log:
Issue #7429 - accessing a not allocated structure causes segfault... (tgrman, fix by myself
inspired by suggested fix).
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=36128&r1=36127&r2=36128&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jun 26 14:07:32 2006
@@ -13333,12 +13333,13 @@
return -1;
}
- if (current.chan2 && sipdebug && option_debug > 3)
- ast_log(LOG_DEBUG, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name);
-
- /* Stop music on hold on this channel */
- if (current.chan2)
+ if (current.chan2) {
+ if (sipdebug && option_debug > 3)
+ ast_log(LOG_DEBUG, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name);
+
+ /* Stop music on hold on this channel */
ast_moh_stop(current.chan2);
+ }
ast_set_flag(&p->flags[0], SIP_GOTREFER);
@@ -13370,15 +13371,13 @@
/* Blind transfers and remote attended xfers */
transmit_response(p, "202 Accepted", req);
- if (current.chan2->name) {
- ast_log(LOG_NOTICE, "chan2->name: %s\n", current.chan2->name);
- pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name);
- }
- if (current.chan1) {
+
+ if (current.chan1 && current.chan2) {
ast_log(LOG_NOTICE, "chan1->name: %s\n", current.chan1->name);
pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name);
}
if (current.chan2) {
+ pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name);
pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", p->refer->refer_to_domain);
pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes");
/* One for the new channel */
@@ -13412,6 +13411,7 @@
ast_channel_unlock(current.chan2);
/* Connect the call */
+
/* FAKE ringing if not attended transfer */
if (!p->refer->attendedtransfer)
transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE);
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