diff --git a/conf/default_context.xml b/conf/default_context.xml
index 86f8be3cd4..bedc6e6bac 100644
--- a/conf/default_context.xml
+++ b/conf/default_context.xml
@@ -1,213 +1,91 @@
-<!-- Valid fields in conditions: -->
-<!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
-<!-- rdnis, destination_number, uuid, source, context, chan_name" -->
-
-<!-- *NOTE* The special context name 'any' will match any context -->
 <context name="default">
-  <extension name="556"> <!-- demo phrases -->
-    <condition field="destination_number" expression="^556$">
-      <action application="answer"/>
-      <action application="sleep" data="1000"/>
-      <action application="phrase" data="spell,${caller_id_name}"/>
-      <action application="phrase" data="spell-phonetic,${caller_id_name}"/>
-      <action application="phrase" data="timespec,12:45:15"/>
-      <action application="phrase" data="saydate,0"/>
-      <action application="phrase" data="msgcount,130"/>
-      <action application="phrase" data="ip-addr,66.250.68.194"/>
-      <action application="phrase" data="saydate,$strepoch(2006-03-23 7:23)"/>
-      <!--<action application="phrase" data="timeleft,3:30"/>-->
+  <!-- 
+  if the calling party is the called party, go to their VM
+  if the calling party is NOT the called party dial the extension 
+  (1000-1019) for 30 seconds and go to voicemail if the 
+  call fails (continue_on_fail=true), otherwise hang up after a 
+  successful bridge (hangup_after-bridge=true) 
+  -->
+  <extension name="Local_Extension">
+    <condition field="destination_number" expression="^(10[01][0-9])$" continue="on-true">
+      <action application="set" data="dialed_ext=$1"/>
     </condition>
-  </extension>
-
-  <extension name="7771">
-    <condition field="destination_number" expression="^7771$">
-      <action application="answer"/>
+    <condition field="destination_number" expression="^${caller_id_number}$">
       <action application="set" data="voicemail_authorized=${sip_authorized}"/>
-      <action application="voicemail" data="check demo $${domain} ${sip_mailbox}"/>
+      <action application="voicemail" data="check default $${domain} ${dialed_ext}"/>
+      <anti-action application="set" data="call_timeout=30000"/>
+      <anti-action application="set" data="hangup_after_bridge=true"/>
+      <anti-actino application="set" data="continue_on_fail=true"/>
+      <anti-action application="bridge" data="sofia/$${domain}/${dialed_ext}"/>
+      <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/>
+    </condition>
+  </extension>                                                                                                                                                 
+ 
+  <!--
+  start a dynamic conference with the settings of the 
+  "default" conference profile in conference.conf.xml
+  -->                                                                                                                                                       
+  <extension name="Conferences">                                                                                                                           
+    <condition field="destination_number" expression="^(3\d{3})$">                                                                                             
+      <action application="conference" data="$1@default"/>                                                                                                     
+    </condition>                                                                                                                                               
+  </extension>           
+ 
+  <!-- dial the freeswitch conference via IAX-->
+  <extension name="FS_Conf_IAX">
+    <condition field="destination_number" expression="^888$">
+      <action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>
     </condition>
   </extension>
-
-  <extension name="7772">
-    <condition field="destination_number" expression="^7772$">
+ 
+  <!--
+  if the number starts with a *393 then strip the *393 
+  and dial the remainder of the number out our "fwd" gateway
+  -->
+  <extension name="FWD">
+    <condition field="destination_number" expression="^\*393([\*\d]+)$">
+    <!-- Just incase they are using Asterisk let's disable VAD and only use G.711 -->
+      <action application="export" data="rtp_disable_vad_out=true"/>
+      <action application="export" data="absolute_codec_string=PCMU@20i,PCMA@20i"/>
+      <action application="bridge" data="sofia/gateway/fwd/$1"/>
+    </condition>
+  </extension> 
+ 
+  <!-- a sample IVR  -->
+  <extension name="IVR_Demo">
+    <condition field="destination_number" expression="5000">
+      <action application="ivr" data="demo"/>
+    </condition>
+  </extension>
+ 
+ 
+  <!-- 
+  Parking extensions... transferring calls to 5900 will park them in a 
+  queue. dialing 5901 will let you pull them out of the queue.
+  -->
+  <extension name="park">
+    <condition field="destination_number" expression="^5900$">
+      <action application="fifo" data="default in undef soundfiles/swimp.raw"/>
+    </condition>
+  </extension>
+ 
+  <extension name="unpark">
+    <condition field="destination_number" expression="^5901$">
       <action application="answer"/>
-      <action application="voicemail" data="demo $${domain} ${sip_mailbox}"/>
+      <action application="fifo" data="default out nowait"/>
     </condition>
-  </extension>
-
-  <extension name="9193">
-    <condition field="destination_number" expression="^9193$">
-      <action application="set" data="bridge_pre_execute_bleg_app=soundtouch"/>
-      <!-- send or recv indicates which direction the dtmf is parsed from 
-	   since this example is send and it's being called on the b leg
-	   the application will intercept the dtmf from being sent to the b leg 
-	   a.k.a. by the dtmf of the A leg.
-	   if it were 'recv' then it would be parsed when the dtmf was 
-	   received *from* the b leg so it could control itself.
-	   The optional keywords "read" and "write" will also change the stream replaced
-      -->
-      <action application="set" data="bridge_pre_execute_bleg_data=send -4s"/>
-      <action application="bridge" data="sofia/$${domain}/foo"/>
-    </condition>
-  </extension>
-
-  <extension name="9192">
-    <condition field="destination_number" expression="^9192$">
-      <!-- Maintain Buffer of 128k of audio (default is 64k) -->
-      <action application="set" data="stream_prebuffer=131072"/>
-      <!-- Play a stream -->
-      <action application="playback" data="shout://mp3.ihets.org/wfyihd132"/>
-    </condition>
-  </extension>
-
-  <!-- FreeNUM.org ISN Lookups -->
-  <extension name="ISN_Lookup">
-    <condition field="destination_number" expression="^([0-9]*)\*([0-9]*)$">
-      <action application="enum" data="$1 $2.freenum.org"/>
-      <action application="log" data="${enum_auto_route}"/>
-      <action application="bridge" data="${enum_auto_route}"/>
-    </condition>
-  </extension>
-
-  <!-- Example extension for require auth per-call. -->
-  <extension name="9191">
-    <!-- Match the destination digits of 9191 -->
-    <condition field="destination_number" expression="^9191$"/>
-    <!-- Make sure the sip_authorized variable is set (set on all authed calls) 
-	 If it isn't, then send an auth challange.
-    -->
-    <condition field="${sip_authorized}" expression="true">
-      <anti-action application="reject" data="407"/>
-    </condition>
-
-    <!-- If you made it here all is well and the call is authed.
-	 Do whatever you wish.
-    -->
+  </extension> 
+ 
+  <!-- 
+  this is a "catchall" extension that will match anything that's 
+  unmatched thus far.... uncomment and reloadxml to activate it
+  -->
+  <!--
+  <extension name="catchall">
     <condition>
-      <action application="playback" data="/tmp/itworked.wav"/>
+      <action application="set" data="effective_callerid_name=another noob"/>
+      <action application="bridge" data="sofia/$${domain}/888@conference.freeswitch.org"/>
     </condition>
   </extension>
-
-  <extension name="tollfree">
-    <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
-      <action application="enum" data="$1"/>
-      <action application="bridge" data="${enum_auto_route}"/>
-    </condition>
-  </extension>
-
-  <!-- Call the FreeSWITCH conference via IAX -->
-  <!--<extension name="FreeSWITCH Conference IAX">-->
-  <!--<condition field="destination_number" expression="^8888$">-->
-  <!--<action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>-->
-  <!--</condition>-->
-  <!--</extension>-->
-
-  <extension name="set_codec" continue="true">
-    <condition field="source" expression="mod_portaudio">
-      <action application="export" data="absolute_codec_string=$${global_codec_prefs}"/>
-      <action application="export" data="nolocal:jitterbuffer_msec=180"/>
-    </condition>
-  </extension>
-
-  <extension name="testmusic">
-    <condition field="destination_number" expression="^1234$">
-      <!-- Request a certain tone/file to be played while you wait for the call to be answered-->
-      <action application="set" data="ringback=${us-ring}"/>
-      <!--<action application="set" data="ringback=/home/ring.wav"/>-->
-      <!--<action application="set" data="jitterbuffer_msec=180"/>-->
-      <action application="bridge" data="sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
-    </condition>
-  </extension>
-
-  <!-- Enter an existing conference -->
-  <extension name="1000">
-    <condition field="destination_number" expression="^1000$">
-      <action application="conference" data="freeswitch"/>
-    </condition>
-  </extension>
-
-  <!-- Start a dynamic conference and call someone at the same time -->
-  <extension name="2000">
-    <condition field="destination_number" expression="^2000$">
-      <action application="conference" data="bridge:mydynaconf:sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
-    </condition>
-  </extension>
-
-  <!-- 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 desired file.  This is for demo purposes both actions -->
-  <!-- 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"/>
-    <condition field="destination_number" expression="(.*)">
-      <action application="bridge" data="wanpipe/pri/a/a/$1"/>
-    </condition>
-  </extension>
-
-  <!-- Push a call into the fifo Queue (Park) -->
-  <extension name="Park Queue">
-    <condition field="destination_number" expression="400">
-      <!-- This will place the call into myqueue and play moh.wav while waiting and announce before bridge -->
-      <action application="fifo" data="in myqueue /tmp/announce.wav /tmp/moh.wav"/>
-      <!-- This is an example with MOH but no announce -->
-      <!-- <action application="fifo" data="in myqueue undef /tmp/moh.wav"/>  -->
-    </condition>
-  </extension>
-
-  <!-- Pop a call out of the fifo Queue (Unpark) -->
-  <extension name="Unpark Queue">
-    <condition field="destination_number" expression="401">
-      <!-- This is "wait" mode which means the caller/agent can press * to go back and wait ending the bridge --> 
-      <action application="fifo" data="out myqueue /tmp/queue-announce.wav /tmp/agent-moh.wav"/>
-      <!-- This is "nowait" which will return to the dial plan if nobody is in queue -->
-      <!-- <action application="fifo" data="out myqueue nowait"/> -->
-    </condition>
-  </extension>
-
-  <!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
-  <extension name="9999">
-    <condition field="source" expression="mod_iax"/>
-    <condition field="destination_number" expression="9999">
-      <action application="playback" data="/var/sounds/beep.gsm"/>
-    </condition>
-  </extension>
-
-  <!--This extension will start a conference and invite several people upon entering -->
-  <extension name="0911">
-    <condition field="destination_number" expression="0911">
-      
-      <!--These params effect the outcalls made once you join-->
-      <action application="set" data="conference_auto_outcall_caller_id_name=pissed off boss"/>
-      <action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
-      <action application="set" data="conference_auto_outcall_timeout=60"/>
-      <action application="set" data="conference_auto_outcall_flags=none"/>
-      <action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>
-
-      <!--Add as many of these as you need, These are the people you are going to call-->
-      <action application="conference_set_auto_outcall" data="sofia/gateway/mygateway/12121231234"/>
-      <action application="conference_set_auto_outcall" data="sofia/$${domain}/1234@somewhere.com"/>
-      
-      <action application="conference" data="cool@default"/>
-    </condition>
-  </extension>
-
+  -->
 </context>
diff --git a/conf/default_context_old.xml b/conf/default_context_old.xml
new file mode 100644
index 0000000000..86f8be3cd4
--- /dev/null
+++ b/conf/default_context_old.xml
@@ -0,0 +1,213 @@
+<!-- Valid fields in conditions: -->
+<!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
+<!-- rdnis, destination_number, uuid, source, context, chan_name" -->
+
+<!-- *NOTE* The special context name 'any' will match any context -->
+<context name="default">
+  <extension name="556"> <!-- demo phrases -->
+    <condition field="destination_number" expression="^556$">
+      <action application="answer"/>
+      <action application="sleep" data="1000"/>
+      <action application="phrase" data="spell,${caller_id_name}"/>
+      <action application="phrase" data="spell-phonetic,${caller_id_name}"/>
+      <action application="phrase" data="timespec,12:45:15"/>
+      <action application="phrase" data="saydate,0"/>
+      <action application="phrase" data="msgcount,130"/>
+      <action application="phrase" data="ip-addr,66.250.68.194"/>
+      <action application="phrase" data="saydate,$strepoch(2006-03-23 7:23)"/>
+      <!--<action application="phrase" data="timeleft,3:30"/>-->
+    </condition>
+  </extension>
+
+  <extension name="7771">
+    <condition field="destination_number" expression="^7771$">
+      <action application="answer"/>
+      <action application="set" data="voicemail_authorized=${sip_authorized}"/>
+      <action application="voicemail" data="check demo $${domain} ${sip_mailbox}"/>
+    </condition>
+  </extension>
+
+  <extension name="7772">
+    <condition field="destination_number" expression="^7772$">
+      <action application="answer"/>
+      <action application="voicemail" data="demo $${domain} ${sip_mailbox}"/>
+    </condition>
+  </extension>
+
+  <extension name="9193">
+    <condition field="destination_number" expression="^9193$">
+      <action application="set" data="bridge_pre_execute_bleg_app=soundtouch"/>
+      <!-- send or recv indicates which direction the dtmf is parsed from 
+	   since this example is send and it's being called on the b leg
+	   the application will intercept the dtmf from being sent to the b leg 
+	   a.k.a. by the dtmf of the A leg.
+	   if it were 'recv' then it would be parsed when the dtmf was 
+	   received *from* the b leg so it could control itself.
+	   The optional keywords "read" and "write" will also change the stream replaced
+      -->
+      <action application="set" data="bridge_pre_execute_bleg_data=send -4s"/>
+      <action application="bridge" data="sofia/$${domain}/foo"/>
+    </condition>
+  </extension>
+
+  <extension name="9192">
+    <condition field="destination_number" expression="^9192$">
+      <!-- Maintain Buffer of 128k of audio (default is 64k) -->
+      <action application="set" data="stream_prebuffer=131072"/>
+      <!-- Play a stream -->
+      <action application="playback" data="shout://mp3.ihets.org/wfyihd132"/>
+    </condition>
+  </extension>
+
+  <!-- FreeNUM.org ISN Lookups -->
+  <extension name="ISN_Lookup">
+    <condition field="destination_number" expression="^([0-9]*)\*([0-9]*)$">
+      <action application="enum" data="$1 $2.freenum.org"/>
+      <action application="log" data="${enum_auto_route}"/>
+      <action application="bridge" data="${enum_auto_route}"/>
+    </condition>
+  </extension>
+
+  <!-- Example extension for require auth per-call. -->
+  <extension name="9191">
+    <!-- Match the destination digits of 9191 -->
+    <condition field="destination_number" expression="^9191$"/>
+    <!-- Make sure the sip_authorized variable is set (set on all authed calls) 
+	 If it isn't, then send an auth challange.
+    -->
+    <condition field="${sip_authorized}" expression="true">
+      <anti-action application="reject" data="407"/>
+    </condition>
+
+    <!-- If you made it here all is well and the call is authed.
+	 Do whatever you wish.
+    -->
+    <condition>
+      <action application="playback" data="/tmp/itworked.wav"/>
+    </condition>
+  </extension>
+
+  <extension name="tollfree">
+    <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
+      <action application="enum" data="$1"/>
+      <action application="bridge" data="${enum_auto_route}"/>
+    </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@conference.freeswitch.org/888"/>-->
+  <!--</condition>-->
+  <!--</extension>-->
+
+  <extension name="set_codec" continue="true">
+    <condition field="source" expression="mod_portaudio">
+      <action application="export" data="absolute_codec_string=$${global_codec_prefs}"/>
+      <action application="export" data="nolocal:jitterbuffer_msec=180"/>
+    </condition>
+  </extension>
+
+  <extension name="testmusic">
+    <condition field="destination_number" expression="^1234$">
+      <!-- Request a certain tone/file to be played while you wait for the call to be answered-->
+      <action application="set" data="ringback=${us-ring}"/>
+      <!--<action application="set" data="ringback=/home/ring.wav"/>-->
+      <!--<action application="set" data="jitterbuffer_msec=180"/>-->
+      <action application="bridge" data="sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
+    </condition>
+  </extension>
+
+  <!-- Enter an existing conference -->
+  <extension name="1000">
+    <condition field="destination_number" expression="^1000$">
+      <action application="conference" data="freeswitch"/>
+    </condition>
+  </extension>
+
+  <!-- Start a dynamic conference and call someone at the same time -->
+  <extension name="2000">
+    <condition field="destination_number" expression="^2000$">
+      <action application="conference" data="bridge:mydynaconf:sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
+    </condition>
+  </extension>
+
+  <!-- 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 desired file.  This is for demo purposes both actions -->
+  <!-- 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"/>
+    <condition field="destination_number" expression="(.*)">
+      <action application="bridge" data="wanpipe/pri/a/a/$1"/>
+    </condition>
+  </extension>
+
+  <!-- Push a call into the fifo Queue (Park) -->
+  <extension name="Park Queue">
+    <condition field="destination_number" expression="400">
+      <!-- This will place the call into myqueue and play moh.wav while waiting and announce before bridge -->
+      <action application="fifo" data="in myqueue /tmp/announce.wav /tmp/moh.wav"/>
+      <!-- This is an example with MOH but no announce -->
+      <!-- <action application="fifo" data="in myqueue undef /tmp/moh.wav"/>  -->
+    </condition>
+  </extension>
+
+  <!-- Pop a call out of the fifo Queue (Unpark) -->
+  <extension name="Unpark Queue">
+    <condition field="destination_number" expression="401">
+      <!-- This is "wait" mode which means the caller/agent can press * to go back and wait ending the bridge --> 
+      <action application="fifo" data="out myqueue /tmp/queue-announce.wav /tmp/agent-moh.wav"/>
+      <!-- This is "nowait" which will return to the dial plan if nobody is in queue -->
+      <!-- <action application="fifo" data="out myqueue nowait"/> -->
+    </condition>
+  </extension>
+
+  <!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
+  <extension name="9999">
+    <condition field="source" expression="mod_iax"/>
+    <condition field="destination_number" expression="9999">
+      <action application="playback" data="/var/sounds/beep.gsm"/>
+    </condition>
+  </extension>
+
+  <!--This extension will start a conference and invite several people upon entering -->
+  <extension name="0911">
+    <condition field="destination_number" expression="0911">
+      
+      <!--These params effect the outcalls made once you join-->
+      <action application="set" data="conference_auto_outcall_caller_id_name=pissed off boss"/>
+      <action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
+      <action application="set" data="conference_auto_outcall_timeout=60"/>
+      <action application="set" data="conference_auto_outcall_flags=none"/>
+      <action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>
+
+      <!--Add as many of these as you need, These are the people you are going to call-->
+      <action application="conference_set_auto_outcall" data="sofia/gateway/mygateway/12121231234"/>
+      <action application="conference_set_auto_outcall" data="sofia/$${domain}/1234@somewhere.com"/>
+      
+      <action application="conference" data="cool@default"/>
+    </condition>
+  </extension>
+
+</context>