more tweaks and docs

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9715 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/dialplan/default.xml

@@ -1,3 +1,14 @@
+<!--
+    NOTICE:
+    
+    This context is usually accessed via authenticated or transfered callers from the public context
+    from the sip profile sitting on port 5060.
+    
+    Authenitcated users will use the user_context variable on the user to determin what context
+    they can access.  You can also add a user in the directory with the cidr= attribute acl.conf.xml
+    will build the domains acl using this value.
+-->
+
 <?xml version="1.0" encoding="utf-8"?>
 <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
 <include>
@@ -82,7 +93,7 @@
     <!-- If to_host and from_host differ this is an external sip URI so lets send it out. -->
     <extension name="external_sip_uri" continue="true">
       <condition field="source" expression="mod_sofia"/>
-      <condition field="${outside_call}" expression="^$"/>
+      <condition field="${outside_call}"a expression="^$"/>
       <condition field="${sip_req_host}" expression="${sip_from_host}">
 	<anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/>
       </condition>
@@ -106,9 +117,6 @@
       </condition>
     </extension>
 
-
-
-
     <extension name="eavesdrop">
       <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
 	<action application="answer"/>
@@ -428,6 +436,11 @@
       </condition>
     </extension>
 
+    <!--
+	You will no longer hear the bong tone.  The wav file is playing stating the call is secure.
+	The file will not play unless you have both TLS and SRTP active.
+    -->
+
     <extension name="hold_music">
       <condition field="destination_number" expression="^9999$"/>
       <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
@@ -440,7 +453,23 @@
       </condition>
     </extension>
 
+    <!--
+	You can place files in the default directory to get included.
+    -->
     <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
+    
+    <!--
+	WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+	
+	Anything you put below this line will usually get ignored to to the file in 
+	default/99999_enum.xml as it will transfer the call to the enum dialplan.
+
+	WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+    -->
+
+    <!--
+	This is an example of how to overide the RURI on an outgoing invite to a registered contact.
+    -->
     <!--
     <extension name="refer">
       <condition field="${sip_refer_to}">
@@ -454,11 +483,7 @@
 	<action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/>
       </condition>
     </extension>
-    -->
+
-    <!--
-	This is an example of how to overide the RURI on an outgoing invite to a registered contact.
-    -->
-    <!--
     <extension name="ruri">
       <condition field="destination_number" expression="^ruri$">
 	<action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
@@ -475,5 +500,7 @@
     </extension>
     -->
 
+    <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! -->
+
   </context>
 </include>

conf/dialplan/public.xml

@@ -1,13 +1,18 @@
+<!--
+    NOTICE:
+
+    This context is usually accessed via the external sip profile sitting on port 5080.
+    
+    It is recommended to have separate inbound and outbound contexts.  Not only for security
+    but clearing up why you would need to do such a thing.  You don't want outside un-authenticated
+    callers hitting your default context which allows dialing calls thru your providers and results 
+    in Toll Fraud.
+-->
+
 <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
 <include>
   <context name="public">
 
-    <extension name="outside_call" continue="true">
-      <condition break="never">
-	<action application="set" data="outside_call=true"/>
-      </condition>
-    </extension>
-
     <extension name="unloop">
       <condition field="${unroll_loops}" expression="^true$"/>
       <condition field="${sip_looped_call}" expression="^true$">
@@ -15,6 +20,17 @@
       </condition>
     </extension>
 
+    <!--
+	Tag anything pass thru here as an outside_call so you can make sure not
+	to create any routing loops based on the conditions that it came from 
+	the outside of the switch.  
+    -->
+    <extension name="outside_call" continue="true">
+      <condition break="never">
+	<action application="set" data="outside_call=true"/>
+      </condition>
+    </extension>
+
     <!-- Try to get target_domain from the sip_req_params variable -->
     <extension name="set_domain" continue="true">
       <condition field="${target_domain}" expression="^$" break="never"/>
@@ -38,6 +54,9 @@
       </condition>
     </extension>
     
+    <!--
+	You can place files in the public directory to get included.
+    -->
     <X-PRE-PROCESS cmd="include" data="public/*.xml"/>
 
   </context>