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
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281651 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/UPGRADE-1.8.txt b/UPGRADE-1.8.txt
index 677fdb7..905681f 100644 (file)
--- a/UPGRADE-1.8.txt
+++ b/UPGRADE-1.8.txt
@@ -20,6 +20,9 @@
 
 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
   option is set to 'no' was modified such that instead of converting '/' to '0',

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 0e207ca..cf3c0da 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -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 */

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 35f4f68..c61e878 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -356,6 +356,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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