[asterisk-commits] branch 1.2 r12495 -
/branches/1.2/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Mar 10 05:13:54 MST 2006
Author: oej
Date: Fri Mar 10 06:13:51 2006
New Revision: 12495
URL: http://svn.digium.com/view/asterisk?rev=12495&view=rev
Log:
Issue #5937 - Make sure SIP CANCEL's are re-transmitted
Modified:
branches/1.2/channels/chan_sip.c
Modified: branches/1.2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?rev=12495&r1=12494&r2=12495&view=diff
==============================================================================
--- branches/1.2/channels/chan_sip.c (original)
+++ branches/1.2/channels/chan_sip.c Fri Mar 10 06:13:51 2006
@@ -2447,13 +2447,15 @@
if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
if (needcancel) { /* Outgoing call, not up */
if (ast_test_flag(p, SIP_OUTGOING)) {
+ /* stop retransmitting an INVITE that has not received a response */
+ __sip_pretend_ack(p);
+
+ /* Send a new request: CANCEL */
transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
ast_clear_flag(&locflags, SIP_NEEDDESTROY);
- sip_scheddestroy(p, 15000);
- /* stop retransmitting an INVITE that has not received a response */
- __sip_pretend_ack(p);
+ sip_scheddestroy(p, 32000);
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
only if the channel is not auto-congested */
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