git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6748 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/sip_profiles/default.xml

@@ -17,7 +17,6 @@
   <settings>
     <param name="debug" value="0"/>
     <param name="sip-trace" value="no"/>
-    <param name="log-level" value="0"/>
     <param name="context" value="public"/>
     <param name="rfc2833-pt" value="101"/>
     <param name="sip-port" value="5060"/>
@@ -62,7 +61,7 @@
     <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec 
 	that the originator is using-->
     <!--<param name="disable-transcoding" value="true"/>-->
-    <param name="auth-calls" value="true"/>
+    <param name="auth-calls" value="false"/>
     <!-- on authed calls, authenticate *all* the packets not just invite -->
     <!-- <param name="auth-all-packets" value="true"/> -->
     

conf/sip_profiles/nat.xml

@@ -7,7 +7,6 @@
   <settings>
     <param name="debug" value="0"/>
     <param name="sip-trace" value="no"/>
-    <param name="log-level" value="0"/>
     <param name="rfc2833-pt" value="101"/>
     <param name="sip-port" value="5070"/>
     <param name="dialplan" value="XML"/>

conf/sip_profiles/outbound.xml

@@ -12,7 +12,6 @@
   <settings>
     <param name="debug" value="0"/>
     <param name="sip-trace" value="no"/>
-    <param name="log-level" value="0"/>
     <param name="rfc2833-pt" value="101"/>
     <param name="sip-port" value="5080"/>
     <param name="dialplan" value="XML"/>