[svn-commits] russell: trunk r281651 - in /trunk: ./ channels/sip/include/ configs/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Aug 10 16:50:30 CDT 2010
Author: russell
Date: Tue Aug 10 16:50:24 2010
New Revision: 281651
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=281651
Log:
Merged revisions 281650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines
Change the default value for alwaysauthreject in sip.conf to "yes".
(closes issue #17756)
Reported by: oej
........
Modified:
trunk/ (props changed)
trunk/UPGRADE-1.8.txt
trunk/channels/sip/include/sip.h
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: trunk/UPGRADE-1.8.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE-1.8.txt?view=diff&rev=281651&r1=281650&r2=281651
==============================================================================
--- trunk/UPGRADE-1.8.txt (original)
+++ trunk/UPGRADE-1.8.txt Tue Aug 10 16:50:24 2010
@@ -19,6 +19,9 @@
===========================================================
From 1.6.2 to 1.8:
+
+* The default value for the alwaysauthreject option in sip.conf has been changed
+ from "no" to "yes".
* The behavior of the 'parkedcallstimeout' has changed slightly. The formulation
of the extension name that a timed out parked call is delivered to when this
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=281651&r1=281650&r2=281651
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Tue Aug 10 16:50:24 2010
@@ -214,7 +214,7 @@
#define DEFAULT_MATCHEXTERNADDRLOCALLY FALSE /*!< Match extern IP locally default setting */
#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
-#define DEFAULT_ALWAYSAUTHREJECT FALSE /*!< Don't reject authentication requests always */
+#define DEFAULT_ALWAYSAUTHREJECT TRUE /*!< Don't reject authentication requests always */
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=281651&r1=281650&r2=281651
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Tue Aug 10 16:50:24 2010
@@ -356,6 +356,7 @@
; instead of letting the requester know whether there was
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.
+ ; This option is set to "yes" by default.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
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