[asterisk-commits] russell: branch 1.8 r281650 - in /branches/1.8: ./ channels/sip/include/ conf...

[SVN commits to the Asterisk project]

Author: russell
Date: Tue Aug 10 16:47:31 2010
New Revision: 281650

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=281650
Log:
Change the default value for alwaysauthreject in sip.conf to "yes".

(closes issue #17756)
Reported by: oej

Modified:
    branches/1.8/UPGRADE.txt
    branches/1.8/channels/sip/include/sip.h
    branches/1.8/configs/sip.conf.sample

Modified: branches/1.8/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/UPGRADE.txt?view=diff&rev=281650&r1=281649&r2=281650
==============================================================================
--- branches/1.8/UPGRADE.txt (original)
+++ branches/1.8/UPGRADE.txt Tue Aug 10 16:47:31 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: branches/1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sip/include/sip.h?view=diff&rev=281650&r1=281649&r2=281650
==============================================================================
--- branches/1.8/channels/sip/include/sip.h (original)
+++ branches/1.8/channels/sip/include/sip.h Tue Aug 10 16:47:31 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: branches/1.8/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?view=diff&rev=281650&r1=281649&r2=281650
==============================================================================
--- branches/1.8/configs/sip.conf.sample (original)
+++ branches/1.8/configs/sip.conf.sample Tue Aug 10 16:47:31 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