[asterisk-commits] bebuild: tag 10.12.3-digiumphones r397827 - in /tags/10.12.3-digiumphones: ./...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Aug 27 14:24:45 CDT 2013
Author: bebuild
Date: Tue Aug 27 14:24:43 2013
New Revision: 397827
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=397827
Log:
Merge changes for 10.12.3-digiumphones
* AST-2013-004
* AST-2013-005
Removed:
tags/10.12.3-digiumphones/asterisk-10.12.2-digiumphones-summary.html
tags/10.12.3-digiumphones/asterisk-10.12.2-digiumphones-summary.txt
Modified:
tags/10.12.3-digiumphones/ (props changed)
tags/10.12.3-digiumphones/.version
tags/10.12.3-digiumphones/ChangeLog
tags/10.12.3-digiumphones/channels/chan_sip.c
Propchange: tags/10.12.3-digiumphones/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Propchange: tags/10.12.3-digiumphones/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Tue Aug 27 14:24:43 2013
@@ -1,2 +1,2 @@
/branches/1.8:378967
-/branches/10:379510,379777,383977,383982
+/branches/10:379510,379777,383977,383982,397711,397757
Modified: tags/10.12.3-digiumphones/.version
URL: http://svnview.digium.com/svn/asterisk/tags/10.12.3-digiumphones/.version?view=diff&rev=397827&r1=397826&r2=397827
==============================================================================
--- tags/10.12.3-digiumphones/.version (original)
+++ tags/10.12.3-digiumphones/.version Tue Aug 27 14:24:43 2013
@@ -1,1 +1,1 @@
-10.12.2-digiumphones
+10.12.3-digiumphones
Modified: tags/10.12.3-digiumphones/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/10.12.3-digiumphones/ChangeLog?view=diff&rev=397827&r1=397826&r2=397827
==============================================================================
--- tags/10.12.3-digiumphones/ChangeLog (original)
+++ tags/10.12.3-digiumphones/ChangeLog Tue Aug 27 14:24:43 2013
@@ -1,3 +1,21 @@
+2013-08-27 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.12.3-digiumphones Released.
+
+ * AST-2013-005: Remote crash from invalid SDP
+
+ If the SIP channel driver processes an invalid SDP that defines media
+ descriptions before connection information, it may attempt to
+ reference the socket address information even though that information
+ has not yet been set. This will cause a crash.
+
+ This patch adds checks when handling the various media descriptions
+ that ensures the media descriptions are handled only if we have
+ connection information suitable for that media.
+
+ Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and
+ providng the solution to this problem.
+
2013-03-27 Asterisk Development Team <asteriskteam at digium.com>
* Asterisk 10.12.2-digiumphones Released.
Modified: tags/10.12.3-digiumphones/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/tags/10.12.3-digiumphones/channels/chan_sip.c?view=diff&rev=397827&r1=397826&r2=397827
==============================================================================
--- tags/10.12.3-digiumphones/channels/chan_sip.c (original)
+++ tags/10.12.3-digiumphones/channels/chan_sip.c Tue Aug 27 14:24:43 2013
@@ -10000,7 +10000,7 @@
/* Setup audio address and port */
if (p->rtp) {
- if (portno > 0) {
+ if (sa && portno > 0) {
ast_sockaddr_set_port(sa, portno);
ast_rtp_instance_set_remote_address(p->rtp, sa);
if (debug) {
@@ -10047,7 +10047,7 @@
/* Setup video address and port */
if (p->vrtp) {
- if (vportno > 0) {
+ if (vsa && vportno > 0) {
ast_sockaddr_set_port(vsa, vportno);
ast_rtp_instance_set_remote_address(p->vrtp, vsa);
if (debug) {
@@ -10064,7 +10064,7 @@
/* Setup text address and port */
if (p->trtp) {
- if (tportno > 0) {
+ if (tsa && tportno > 0) {
ast_sockaddr_set_port(tsa, tportno);
ast_rtp_instance_set_remote_address(p->trtp, tsa);
if (debug) {
@@ -10087,7 +10087,7 @@
/* Setup image address and port */
if (p->udptl) {
- if (udptlportno > 0) {
+ if (isa && udptlportno > 0) {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
ast_rtp_instance_get_remote_address(p->rtp, isa);
if (!ast_sockaddr_isnull(isa) && debug) {
@@ -26915,7 +26915,7 @@
p->invitestate = INV_TERMINATED;
p->pendinginvite = 0;
acked = __sip_ack(p, seqno, 1 /* response */, 0);
- if (find_sdp(req)) {
+ if (p->owner && find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_NONE)) {
return -1;
}
More information about the asterisk-commits
mailing list