reformat sample conf, remove exosip references

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2906 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/freeswitch.xml

@@ -1,12 +1,15 @@
 <?xml version="1.0"?>
 <document type="freeswitch/xml">
+
   <section name="configuration" description="Various Configuration">
+    
     <configuration name="switch.conf" description="Modules">
       <settings>
         <!--Most channels to allow at once -->
         <param name="max-sessions" value="1000"/>
       </settings>
     </configuration>
+
     <configuration name="modules.conf" description="Modules">
       <modules>
         <!-- Loggers (I'd load these first) -->
@@ -17,46 +20,46 @@
         <!-- <load module="mod_xml_rpc"/> -->
 
         <!-- Event Handlers -->
-	<!-- <load module="mod_event_multicast"/> -->
-	<!-- <load module="mod_event_test"/> -->
-	<!-- <load module="mod_zeroconf"/> -->
-	<!-- <load module="mod_xmpp_event"/> -->
-	<!-- <load module="mod_event_socket"/> -->
         <!-- <load module="mod_cdr"/> -->
+        <!-- <load module="mod_event_multicast"/> -->
+        <!-- <load module="mod_event_socket"/> -->
+        <!-- <load module="mod_xmpp_event"/> -->
+        <!-- <load module="mod_zeroconf"/> -->
 
         <!-- Directory Interfaces -->
         <!-- <load module="mod_ldap"/> -->
 
         <!-- Endpoints -->
-	<load module="mod_sofia"/>
+        <!-- <load module="mod_dingaling"/> -->
         <!--<load module="mod_iax"/>-->
         <load module="mod_portaudio"/>
-	<!-- <load module="mod_woomera"/> -->
+        <load module="mod_sofia"/>
         <!-- <load module="mod_wanpipe"/> -->
-	<!-- <load module="mod_dingaling"/> -->
+        <!-- <load module="mod_woomera"/> -->
 
         <!-- Applications -->
         <load module="mod_bridgecall"/>
-	<load module="mod_echo"/>
-	<load module="mod_dptools"/>
-	<!-- <load module="mod_ivrtest"/> -->
-	<load module="mod_playback"/>
         <load module="mod_commands"/>
-	<!-- <load module="mod_commands"/> -->
+        <!--<load module="mod_conference"/>-->
+        <load module="mod_dptools"/>
+        <load module="mod_echo"/>
+        <!--<load module="mod_park"/>-->
+        <load module="mod_playback"/>
 
         <!-- Dialplan Interfaces -->
-	<load module="mod_dialplan_xml"/>
         <!-- <load module="mod_dialplan_directory"/> -->
+        <load module="mod_dialplan_xml"/>
 
         <!-- Codec Interfaces -->
         <load module="mod_g711"/>
         <load module="mod_gsm"/>
+        <!-- <load module="mod_ilbc"/> -->
         <load module="mod_l16"/>
         <!-- <load module="mod_speex"/> -->
-	<!-- <load module="mod_ilbc"/> -->
 
         <!-- File Format Interfaces -->
         <load module="mod_sndfile"/>
+        <load module="mod_native_file"/>
 
         <!-- Timers -->
         <load module="mod_softtimer"/>
@@ -69,9 +72,6 @@
         <!-- <load module="mod_cepstral"/> -->
         <!-- <load module="mod_rss"/> -->
 
-	<!-- Conference Bridges -->
-	<!--<load module="mod_conference"/>-->
-
       </modules>
     </configuration>
 
@@ -111,6 +111,7 @@
         <param name="all" value="DEBUG"/>
       </mappings>
     </configuration>
+
     <configuration name="sofia.conf" description="sofia Endpoint">
       <profiles>
         <profile name="test">
@@ -159,6 +160,7 @@
         </profile>
       </profiles>
     </configuration>
+
     <configuration name="syslog.conf" description="Syslog Logger">
       <!-- SYSLOG -->
       <!-- emerg   - system is unusable  -->
@@ -278,8 +280,7 @@
         <param name="auto-login" value="true"/>
         <!-- SASL "plain" or "md5" -->
         <param name="sasl" value="plain"/>
-	<!-- if the server where the jabber is hosted is not the same
+        <!-- if the server where the jabber is hosted is not the same as the one in the jid -->
-	     as the one in the jid -->
         <!--<param name="server" value="alternate.server.com"/>-->
         <!-- Enable TLS or not -->
         <param name="tls" value="true"/>
@@ -296,6 +297,7 @@
         <param name="vad" value="both"/>
       </interface>
     </configuration>
+
     <configuration name="xml_rpc.conf" description="XML RPC">
       <settings>
         <!-- The port where you want to run the http service (default 8080) -->
@@ -304,9 +306,8 @@
         <param name="auth-realm" value="freeswitch"/>
         <param name="auth-user" value="freeswitch"/>
         <param name="auth-pass" value="works"/>
-	<!-- The url to a gateway cgi that can generate xml similar to
+        <!-- The url to a gateway cgi that can generate xml similar to what's in -->
-	     what's in this file only on-the-fly (leave it commented if you dont
+        <!-- this file only on-the-fly (leave it commented if you dont need it)-->
-	     need it) -->
         <!-- one or more |-delim of configuration|directory|dialplan -->
         <!-- <param name="gateway-url" value="http://www.server.com/gateway.cgi" bindings="configuration"/> -->
       </settings>
@@ -320,8 +321,7 @@
       </feeds>
     </configuration>
 
-    <!-- None of these paths are real if you want any of these options
+    <!-- None of these paths are real if you want any of these options you need to really set them up -->
-	 you need to really set them up -->
     <configuration name="conference.conf" description="Audio Conference">
       <!-- Profiles are collections of settings you can reference by name. -->
 
@@ -338,8 +338,8 @@
           <!-- TTS Voice to use -->
           <!--<param name="tts-voice" value="david"/>-->
 
-	  <!-- If TTS is enabled all audio-file params not beginning with '/'
+          <!-- If TTS is enabled all audio-file params not beginning with -->
-	       will be considered text to say with TTS -->
+          <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
 
           <!-- File to play to acknowledge succees -->
           <!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
@@ -373,29 +373,37 @@
       </profiles>
     </configuration>
   </section>
+  
   <section name="dialplan" description="Regex/XML Dialplan">
-    <!-- Valid fields in conditions:
+    <!-- Valid fields in conditions: -->
-	 "dialplan, caller_id_name, ani, ani2, caller_id_number,
+    <!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
-	 network_addr, rdnis, destination_number, uuid, source,
+    <!-- rdnis, destination_number, uuid, source, context, chan_name" -->
-	 context, chan_name" -->
 
     <!-- *NOTE* The special context name 'any' will match any context -->
     <context name="default">
       <extension name="tollfree">
         <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
-	  <action application="bridge" data="exosip/$1-freeswitch@voip.trxtel.com"/>
+          <action application="bridge" data="sofia/test/$1-freeswitch@voip.trxtel.com"/>
         </condition>
       </extension>
 
-      <!--<extension name="devconf">
+      <!-- Call the FreeSWITCH conference via SIP -->
-	  <condition field="destination_number" expression="^888$">
+        <!--<extension name="FreeSWITCH Conference SIP">-->
-	  <action application="bridge" data="exosip/888@66.250.68.194"/>
+          <!--<condition field="destination_number" expression="^888$">-->
-	  </condition>
+            <!--<action application="bridge" data="sofia/test/888@66.250.68.194"/>-->
-	  </extension> -->
+          <!--</condition>-->
+        <!--</extension> -->
+
+      <!-- Call the FreeSWITCH conference via IAX -->
+        <!--<extension name="FreeSWITCH Conference IAX">-->
+          <!--<condition field="destination_number" expression="^8888$">-->
+            <!--<action application="bridge" data="iax/guest@66.250.68.194/888"/>-->
+          <!--</condition>-->
+        <!--</extension>-->
 
       <extension name="testmusic">
         <condition field="destination_number" expression="^1234$">
-	  <action application="bridge" data="exosip/1234@66.250.68.194"/>
+          <action application="bridge" data="sofia/test/1234@66.250.68.194"/>
         </condition>
       </extension>
 
@@ -409,37 +417,33 @@
       <!-- Start a dynamic conference and call someone at the same time -->
       <extension name="2000">
         <condition field="destination_number" expression="^2000$">
-	  <action application="conference" data="bridge:mydynaconf:exosip/1234@66.250.68.194"/>
+          <action application="conference" data="bridge:mydynaconf:sofia/test/1234@66.250.68.194"/>
         </condition>
       </extension>
 
-      <!-- if the destination is an exact match on the extension name
+      <!-- extensions starting with 4, all the numbers after 4 form a numeric filename -->
-	   you do not need any regex in the condition 
+      <!-- continue="true" means keep looking for more extensions to match -->
-	   <extension name="999">
+      <!-- *NOTE* The entire dialplan is parsed ONCE when the call starts -->
-	   <condition><action application="bridge" data="exosip/888@66.250.68.194"/></condition>
+      <!-- so any call info acquired after the various actions cannot -->
-	   </extension>-->
+      <!-- be taken into consideration. -->
-      <!-- extensions starting with 4, all the numbers after 4 form a numeric filename
-	   continue=true means keep looking for more extensions to match
-	   *NOTE* The entire dialplan is parsed ONCE when the call starts 
-	   so any call info acquired after the various actions cannot
-	   be taken into consideration. 
 
-The first match will play a beep and the second one plays
+      <!-- The first match will play a beep and the second one plays -->
-the desired file.  This is for demo purposes both actions
+      <!-- the desired file.  This is for demo purposes both actions -->
-could have been under the same <extension> tag as well.
+      <!-- could have been under the same <extension> tag as well. -->
-      -->
       <extension name="playsound1" continue="true">
         <condition field="source" expression="mod_sofia"/>
         <condition field="destination_number" expression="^4(\d+)">
           <action application="playback" data="/var/sounds/beep.gsm"/>
         </condition>
       </extension>
+
       <extension name="playsound2">
         <condition field="source" expression="mod_sofia"/>
         <condition field="destination_number" expression="^4(\d+)">
           <action application="playback" data="/root/$1.raw"/>
         </condition>
       </extension>
+
       <!-- send everything with a certian RDNIS to Wanpipe ISDN -->
       <extension name="To PRI">
         <condition field="rdnis" expression="8881231234"/>
@@ -447,6 +451,7 @@ could have been under the same <extension> tag as well.
           <action application="bridge" data="wanpipe/a/a/$1"/>
         </condition>
       </extension>
+
       <!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
       <extension name="9999">
         <condition field="source" expression="mod_iax"/>
@@ -454,18 +459,7 @@ could have been under the same <extension> tag as well.
           <action application="playback" data="/var/sounds/beep.gsm"/>
         </condition>
       </extension>
-      <!-- Call the FreeSWITCH conference via SIP -->
+
-      <extension name="FreeSWITCH Conference SIP">
-	<condition field="destination_number" expression="^888$">
-	  <action application="bridge" data="exosip/888@66.250.68.194"/>
-	</condition>
-      </extension>
-      <!-- Call the FreeSWITCH conference via IAX -->
-      <extension name="FreeSWITCH Conference IAX">
-	<condition field="destination_number" expression="^8888$">
-	  <action application="bridge" data="iax/guest@66.250.68.194/888"/>
-	</condition>
-      </extension>
     </context>
   </section>