broken phones.. you know who you are... Please fix it...

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6927 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/sip_profiles/default.xml

@@ -73,10 +73,11 @@
     <!--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"/>-->
+    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
+    <param name="NDLB-broken-auth-hash" value="true"/>
     <param name="auth-calls" value="true"/>
     <!-- on authed calls, authenticate *all* the packets not just invite -->
-    <!-- <param name="auth-all-packets" value="true"/> -->
-    
+    <param name="auth-all-packets" value="true"/>
     <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
     <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
     <!-- rtp inactivity timeout -->

conf/sip_profiles/nat.xml

@@ -20,6 +20,7 @@
     <param name="inbound-codec-negotiation" value="generous"/>
     <param name="nonce-ttl" value="60"/>
     <param name="auth-calls" value="true"/>
+    <param name="NDLB-broken-auth-hash" value="true"/>
     <param name="rtp-timeout-sec" value="1800"/>
     <param name="rtp-ip" value="$${local_ip_v4}"/>
     <param name="sip-ip" value="$${local_ip_v4}"/>

conf/sip_profiles/outbound.xml

@@ -25,6 +25,7 @@
     <param name="inbound-codec-negotiation" value="generous"/>
     <param name="nonce-ttl" value="60"/>
     <param name="auth-calls" value="false"/>
+    <param name="NDLB-broken-auth-hash" value="true"/>
     <param name="rtp-timeout-sec" value="1800"/>
     <param name="rtp-ip" value="$${local_ip_v4}"/>
     <param name="sip-ip" value="$${local_ip_v4}"/>