From: Anthony Minessale <anthony.minessale@gmail.com>
Date: Mon, 11 Dec 2006 19:02:14 +0000 (+0000)
Subject: formatting
X-Git-Tag: v1.0-beta1~1599
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=5daa2313a6e1eec445dd79cecc88121e406707b1;p=thirdparty%2Ffreeswitch.git

formatting

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3605 d0543943-73ff-0310-b7d9-9358b9ac24b2
---

diff --git a/conf/freeswitch.xml b/conf/freeswitch.xml
index bfe1a586a1..6ce1b830a3 100644
--- a/conf/freeswitch.xml
+++ b/conf/freeswitch.xml
@@ -5,8 +5,8 @@
     
     <configuration name="switch.conf" description="Modules">
       <settings>
-        <!--Most channels to allow at once -->
-        <param name="max-sessions" value="1000"/>
+	<!--Most channels to allow at once -->
+	<param name="max-sessions" value="1000"/>
       </settings>
       <!--Any variables defined here will be available in every channel, in the dialplan etc -->
       <variables>
@@ -18,70 +18,70 @@
 
     <configuration name="modules.conf" description="Modules">
       <modules>
-        <!-- Loggers (I'd load these first) -->
-        <load module="mod_console"/>
-        <!-- <load module="mod_syslog"/> -->
+	<!-- Loggers (I'd load these first) -->
+	<load module="mod_console"/>
+	<!-- <load module="mod_syslog"/> -->
 
-        <!-- Multi-Faceted -->
+	<!-- Multi-Faceted -->
 	<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
-        <load module="mod_enum"/>
-
-        <!-- XML Interfaces -->
-        <!-- <load module="mod_xml_rpc"/> -->
-        <!-- <load module="mod_xml_curl"/> -->
-
-        <!-- Event Handlers -->
-        <!-- <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_dingaling"/> -->
-        <!--<load module="mod_iax"/>-->
-        <load module="mod_portaudio"/>
-        <load module="mod_sofia"/>
-        <!-- <load module="mod_wanpipe"/> -->
-        <!-- <load module="mod_woomera"/> -->
-
-        <!-- Applications -->
-        <load module="mod_bridgecall"/>
-        <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_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"/> -->
-
-        <!-- File Format Interfaces -->
-        <load module="mod_sndfile"/>
-        <load module="mod_native_file"/>
-
-        <!-- Timers -->
-        <load module="mod_softtimer"/>
-
-        <!-- Languages -->
-        <!-- <load module="mod_spidermonkey"/> -->
-        <!-- <load module="mod_perl"/> -->
-
-        <!-- ASR /TTS -->
-        <!-- <load module="mod_cepstral"/> -->
-        <!-- <load module="mod_rss"/> -->
+	<load module="mod_enum"/>
+
+	<!-- XML Interfaces -->
+	<!-- <load module="mod_xml_rpc"/> -->
+	<!-- <load module="mod_xml_curl"/> -->
+
+	<!-- Event Handlers -->
+	<!-- <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_dingaling"/> -->
+	<!--<load module="mod_iax"/>-->
+	<load module="mod_portaudio"/>
+	<load module="mod_sofia"/>
+	<!-- <load module="mod_wanpipe"/> -->
+	<!-- <load module="mod_woomera"/> -->
+
+	<!-- Applications -->
+	<load module="mod_bridgecall"/>
+	<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_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"/> -->
+
+	<!-- File Format Interfaces -->
+	<load module="mod_sndfile"/>
+	<load module="mod_native_file"/>
+
+	<!-- Timers -->
+	<load module="mod_softtimer"/>
+
+	<!-- Languages -->
+	<!-- <load module="mod_spidermonkey"/> -->
+	<!-- <load module="mod_perl"/> -->
+
+	<!-- ASR /TTS -->
+	<!-- <load module="mod_cepstral"/> -->
+	<!-- <load module="mod_rss"/> -->
       </modules>
     </configuration>
 
@@ -95,29 +95,29 @@
 
     <configuration name="event_multicast.conf" description="Multicast Event">
       <settings>
-        <param name="address" value="225.1.1.1"/>
-        <param name="port" value="4242"/>
-        <param name="bindings" value="all"/>
+	<param name="address" value="225.1.1.1"/>
+	<param name="port" value="4242"/>
+	<param name="bindings" value="all"/>
       </settings>
     </configuration>
 
     <configuration name="event_socket.conf" description="Socket Client">
       <settings>
-        <param name="listen-ip" value="127.0.0.1"/>
-        <param name="listen-port" value="8021"/>
-        <param name="password" value="ClueCon"/>
+	<param name="listen-ip" value="127.0.0.1"/>
+	<param name="listen-port" value="8021"/>
+	<param name="password" value="ClueCon"/>
       </settings>
     </configuration>
 
     <configuration name="iax.conf" description="IAX Configuration">
       <settings>
-        <param name="debug" value="0"/>
-        <!-- <param name="ip" value="1.2.3.4"> -->
-        <param name="port" value="4569"/>
-        <param name="dialplan" value="XML"/>
-        <param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
-        <param name="codec-master" value="us"/>
-        <param name="codec-rates" value="8"/>
+	<param name="debug" value="0"/>
+	<!-- <param name="ip" value="1.2.3.4"> -->
+	<param name="port" value="4569"/>
+	<param name="dialplan" value="XML"/>
+	<param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
+	<param name="codec-master" value="us"/>
+	<param name="codec-rates" value="8"/>
       </settings>
     </configuration>
 
@@ -125,57 +125,57 @@
       <!-- pick a file name, a function name or 'all' -->
       <!-- map as many as you need for specific debugging -->
       <mappings>
-        <!-- <param name="log_event" value="DEBUG"/> -->
-        <param name="all" value="DEBUG"/>
+	<!-- <param name="log_event" value="DEBUG"/> -->
+	<param name="all" value="DEBUG"/>
       </mappings>
     </configuration>
 
     <configuration name="sofia.conf" description="sofia Endpoint">
       <profiles>
-        <profile name="mydomain1.com">
-          <registrations>
-            <!-- <registration name="asterlink">
-              <param name="register-scheme" value="Digest"/>
-              <param name="register-realm" value=""/>
-              <param name="register-username" value="1001"/>
-              <param name="register-password" value="nhy65tgb"/>
-              <param name="register-from" value="sip:1001@208.64.200.40"/>
-              <param name="register-to" value="sip:1001@conference.freeswitch.org"/>
-              <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
-              <param name="register-frequency" value="20"/>
-            </registration> -->
-          </registrations>
-          <settings>
-            <param name="debug" value="1"/>
-            <param name="rfc2833-pt" value="101"/>
-            <param name="sip-port" value="5060"/>
-            <param name="dialplan" value="XML"/>
-            <param name="dtmf-duration" value="100"/>
-            <param name="codec-prefs" value="PCMU@20i"/>
-            <param name="codec-ms" value="20"/>
-            <param name="use-rtp-timer" value="true"/>
-            <param name="rtp-timer-name" value="soft"/>
-            <param name="rtp-ip" value="192.168.1.20"/>
-            <param name="sip-ip" value="mydomain1.com"/>
-
-            <!-- this lets anything register -->
-            <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
-            <param name="accept-blind-reg" 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"/>-->
-
-            <!-- optional ; -->
-            <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
-            <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
-            <!-- VAD choose one (out is a good choice); -->
-            <!-- <param name="vad" value="in"/> -->
-            <!-- <param name="vad" value="out"/> -->
-            <!-- <param name="vad" value="both"/> -->
-            <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
-          </settings>
-        </profile>
+	<profile name="mydomain1.com">
+	  <registrations>
+	    <!-- <registration name="asterlink">
+		 <param name="register-scheme" value="Digest"/>
+		 <param name="register-realm" value=""/>
+		 <param name="register-username" value="1001"/>
+		 <param name="register-password" value="nhy65tgb"/>
+		 <param name="register-from" value="sip:1001@208.64.200.40"/>
+		 <param name="register-to" value="sip:1001@conference.freeswitch.org"/>
+		 <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
+		 <param name="register-frequency" value="20"/>
+		 </registration> -->
+	  </registrations>
+	  <settings>
+	    <param name="debug" value="1"/>
+	    <param name="rfc2833-pt" value="101"/>
+	    <param name="sip-port" value="5060"/>
+	    <param name="dialplan" value="XML"/>
+	    <param name="dtmf-duration" value="100"/>
+	    <param name="codec-prefs" value="PCMU@20i"/>
+	    <param name="codec-ms" value="20"/>
+	    <param name="use-rtp-timer" value="true"/>
+	    <param name="rtp-timer-name" value="soft"/>
+	    <param name="rtp-ip" value="192.168.1.20"/>
+	    <param name="sip-ip" value="mydomain1.com"/>
+
+	    <!-- this lets anything register -->
+	    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
+	    <param name="accept-blind-reg" 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"/>-->
+
+	    <!-- optional ; -->
+	    <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
+	    <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
+	    <!-- VAD choose one (out is a good choice); -->
+	    <!-- <param name="vad" value="in"/> -->
+	    <!-- <param name="vad" value="out"/> -->
+	    <!-- <param name="vad" value="both"/> -->
+	    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
+	  </settings>
+	</profile>
       </profiles>
     </configuration>
 
@@ -190,147 +190,147 @@
       <!-- info    - informational message  -->
       <!-- debug   - debug-level message -->
       <settings>
-        <param name="ident" value="freeswitch"/>
-        <param name="facility" value="user"/>
-        <param name="format" value="${time} - ${message}"/>
-        <param name="level" value="debug,info,warning-alert"/>
+	<param name="ident" value="freeswitch"/>
+	<param name="facility" value="user"/>
+	<param name="format" value="${time} - ${message}"/>
+	<param name="level" value="debug,info,warning-alert"/>
       </settings>
     </configuration>
 
     <configuration name="woomera.conf" description="Woomera Endpoint">
       <settings>
-        <param name="debug" value="0"/>
+	<param name="debug" value="0"/>
       </settings>
       <interface>
-        <param name="host" value="localhost"/>
-        <param name="port" value="42420"/>
-        <param name="audio-ip" value="127.0.0.1"/>
-        <param name="dialplan" value="XML"/>
+	<param name="host" value="localhost"/>
+	<param name="port" value="42420"/>
+	<param name="audio-ip" value="127.0.0.1"/>
+	<param name="dialplan" value="XML"/>
       </interface>
     </configuration>
 
     <configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
       <settings>
-        <param name="debug" value="1"/>
-        <param name="dialplan" value="XML"/>
-        <param name="mtu" value="320"/>
-        <param name="dtmf-on" value="800"/>
-        <param name="dtmf-off" value="100"/>
-        <param name="supress-dtmf-tone" value="yes"/>
+	<param name="debug" value="1"/>
+	<param name="dialplan" value="XML"/>
+	<param name="mtu" value="320"/>
+	<param name="dtmf-on" value="800"/>
+	<param name="dtmf-off" value="100"/>
+	<param name="supress-dtmf-tone" value="yes"/>
       </settings>
       <span>
-        <param name="span" value="1"/>
-        <param name="node" value="cpe"/>
-        <!-- <param name="switch" value="ni2"/> -->
-        <param name="switch" value="dms100"/>
-        <!-- <param name="switch" value="lucent5e"/> -->
-        <!-- <param name="switch" value="att4ess"/> -->
-        <!-- <param name="switch" value="euroisdn"/> -->
-        <!-- <param name="switch" value="gr303eoc"/> -->
-        <!-- <param name="switch" value="gr303tmc"/> -->
-        <param name="dp" value="national"/>
-        <!-- <param name="dp" value="international"/> -->
-        <!-- <param name="dp" value="local"/> -->
-        <!-- <param name="dp" value="private"/> -->
-        <!-- <param name="dp" value="unknown"/> -->
-        <param name="l1" value="ulaw"/>
-        <!-- <param name="l1" value="alaw"/> -->
-        <param name="bchan" value="1-23"/>
-        <param name="dchan" value="24"/>
-        <param name="dialplan" value="XML"/>
+	<param name="span" value="1"/>
+	<param name="node" value="cpe"/>
+	<!-- <param name="switch" value="ni2"/> -->
+	<param name="switch" value="dms100"/>
+	<!-- <param name="switch" value="lucent5e"/> -->
+	<!-- <param name="switch" value="att4ess"/> -->
+	<!-- <param name="switch" value="euroisdn"/> -->
+	<!-- <param name="switch" value="gr303eoc"/> -->
+	<!-- <param name="switch" value="gr303tmc"/> -->
+	<param name="dp" value="national"/>
+	<!-- <param name="dp" value="international"/> -->
+	<!-- <param name="dp" value="local"/> -->
+	<!-- <param name="dp" value="private"/> -->
+	<!-- <param name="dp" value="unknown"/> -->
+	<param name="l1" value="ulaw"/>
+	<!-- <param name="l1" value="alaw"/> -->
+	<param name="bchan" value="1-23"/>
+	<param name="dchan" value="24"/>
+	<param name="dialplan" value="XML"/>
       </span>
     </configuration>
 
     <configuration name="portaudio.conf" description="Soundcard Endpoint">
       <settings>
-        <param name="debug" value="2"/>
-        <param name="dialplan" value="XML"/>
+	<param name="debug" value="2"/>
+	<param name="dialplan" value="XML"/>
 
-        <!-- partial string match on something in the name or the device # -->
-        <param name="indev" value="USB"/>
-        <param name="outdev" value="USB"/>
+	<!-- partial string match on something in the name or the device # -->
+	<param name="indev" value="USB"/>
+	<param name="outdev" value="USB"/>
 
-        <param name="cid-name" value="FreeSwitch"/>
-        <param name="cid-num" value="5555551212"/>
+	<param name="cid-name" value="FreeSwitch"/>
+	<param name="cid-num" value="5555551212"/>
       </settings>
     </configuration>
 
     <configuration name="zeroconf.conf" description="Zeroconf Event Handler">
       <settings>
-        <param name="publish" value="yes"/>
-        <param name="browse" value="_sip._udp"/>
+	<param name="publish" value="yes"/>
+	<param name="browse" value="_sip._udp"/>
       </settings>
     </configuration>
 
     <configuration name="xmpp_event.conf" description="XMPP Event Handler">
       <settings>
-        <param name="#debug" value="1"/>
-        <param name="jid" value="freeswitch@my.jabber.com/me"/>
-        <param name="passwd" value="mypass"/>
-        <param name="target-jid" value="freeswitch@reader.org/him"/>
+	<param name="#debug" value="1"/>
+	<param name="jid" value="freeswitch@my.jabber.com/me"/>
+	<param name="passwd" value="mypass"/>
+	<param name="target-jid" value="freeswitch@reader.org/him"/>
       </settings>
     </configuration>
 
     <configuration name="dialplan_directory.conf" description="Dialplan Directory">
       <settings>
-        <param name="directory-name" value="ldap"/>
-        <param name="host" value="ldap.freeswitch.org"/>
-        <param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
-        <param name="pass" value="test"/>
-        <param name="base" value="dc=freeswitch,dc=org"/>
+	<param name="directory-name" value="ldap"/>
+	<param name="host" value="ldap.freeswitch.org"/>
+	<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
+	<param name="pass" value="test"/>
+	<param name="base" value="dc=freeswitch,dc=org"/>
       </settings>
     </configuration>
 
     <configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
       <settings>
-        <param name="debug" value="0"/>
-        <param name="codec-prefs" value="PCMU"/>
+	<param name="debug" value="0"/>
+	<param name="codec-prefs" value="PCMU"/>
       </settings>
 
       <!-- *NOTE* change <x-profile></x-profile> to <profile></profile> to enable -->
 
       <!-- Client Profile (Original mode) -->
       <x-profile type="client">
-        <param name="name" value="mydomain.com"/>
-        <param name="login" value="myjid@myserver.com/talk"/>
-        <param name="password" value="mypass"/>
-        <param name="dialplan" value="XML"/>
-        <param name="message" value="Jingle all the way"/>
-        <param name="rtp-ip" value="10.0.0.1"/>
-        <param name="auto-login" value="true"/>
+	<param name="name" value="mydomain.com"/>
+	<param name="login" value="myjid@myserver.com/talk"/>
+	<param name="password" value="mypass"/>
+	<param name="dialplan" value="XML"/>
+	<param name="message" value="Jingle all the way"/>
+	<param name="rtp-ip" value="10.0.0.1"/>
+	<param name="auto-login" value="true"/>
 	<param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/>
-        <!-- SASL "plain" or "md5" -->
-        <param name="sasl" value="plain"/>
-        <!-- if the server where the jabber is hosted is not the same as the one in the jid -->
-        <!--<param name="server" value="alternate.server.com"/>-->
-        <!-- Enable TLS or not -->
-        <param name="tls" value="true"/>
-        <!-- disable to trade async for more calls -->
-        <param name="use-rtp-timer" value="true"/>
-        <!-- or -->
-        <!-- <param name="rtp-ip" value="my_lan_ip"/> -->
-        <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
-        <!-- default extension (if one cannot be determined) -->
-        <param name="exten" value="888"/>
-        <!-- VAD choose one -->
-        <!-- <param name="vad" value="in"/> -->
-        <!-- <param name="vad" value="out"/> -->
-        <param name="vad" value="both"/>
+	<!-- SASL "plain" or "md5" -->
+	<param name="sasl" value="plain"/>
+	<!-- if the server where the jabber is hosted is not the same as the one in the jid -->
+	<!--<param name="server" value="alternate.server.com"/>-->
+	<!-- Enable TLS or not -->
+	<param name="tls" value="true"/>
+	<!-- disable to trade async for more calls -->
+	<param name="use-rtp-timer" value="true"/>
+	<!-- or -->
+	<!-- <param name="rtp-ip" value="my_lan_ip"/> -->
+	<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
+	<!-- default extension (if one cannot be determined) -->
+	<param name="exten" value="888"/>
+	<!-- VAD choose one -->
+	<!-- <param name="vad" value="in"/> -->
+	<!-- <param name="vad" value="out"/> -->
+	<param name="vad" value="both"/>
       </x-profile>
 
       <!-- Component (Server to Server Login) -->
       <x-profile type="component">
 	<!-- All traffic for *@sub.mydomain.com will come to you -->
 	<param name="name" value="sub.mydomain.com"/>
-        <param name="password" value="secret"/>
+	<param name="password" value="secret"/>
 	<param name="dialplan" value="XML"/>
 	<param name="rtp-ip" value="208.64.200.42"/>
 	<param name="server" value="jabber.server.org:5347"/>
-        <!-- disable to trade async for more calls -->
-        <param name="use-rtp-timer" value="true"/>
+	<!-- disable to trade async for more calls -->
+	<param name="use-rtp-timer" value="true"/>
 	<!-- "_auto_" means the extension will be automaticly set to the called jid -->
-        <param name="exten" value="_auto_"/>
-        <!--<param name="vad" value="both"/>-->
+	<param name="exten" value="_auto_"/>
+	<!--<param name="vad" value="both"/>-->
       </x-profile>
 
     </configuration>
@@ -347,20 +347,20 @@
 
     <configuration name="xml_rpc.conf" description="XML RPC">
       <settings>
-        <!-- The port where you want to run the http service (default 8080) -->
-        <param name="http-port" value="8080"/>
-        <!-- if all 3 of the following params exist all http traffic will require auth -->
-        <param name="auth-realm" value="freeswitch"/>
-        <param name="auth-user" value="freeswitch"/>
-        <param name="auth-pass" value="works"/>
+	<!-- The port where you want to run the http service (default 8080) -->
+	<param name="http-port" value="8080"/>
+	<!-- if all 3 of the following params exist all http traffic will require auth -->
+	<param name="auth-realm" value="freeswitch"/>
+	<param name="auth-user" value="freeswitch"/>
+	<param name="auth-pass" value="works"/>
       </settings>
     </configuration>
 
     <configuration name="rss.conf" description="RSS Parser">
       <feeds>
-        <!-- Just download the files to wherever and refer to them here -->
-        <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
-        <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
+	<!-- Just download the files to wherever and refer to them here -->
+	<!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
+	<!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
       </feeds>
     </configuration>
 
@@ -369,87 +369,103 @@
       <!-- Profiles are collections of settings you can reference by name. -->
 
       <profiles>
-        <profile name="default">
-          <!-- Sample Rate-->
-          <param name="rate" value="8000"/>
-          <!-- Number of milliseconds per frame -->
-          <param name="interval" value="20"/>
-          <!-- Energy level required for audio to be sent to the other users -->
-          <param name="energy-level" value="300"/>
-          <!-- TTS Engine to use -->
-          <!--<param name="tts-engine" value="cepstral"/>-->
-          <!-- TTS Voice to use -->
-          <!--<param name="tts-voice" value="david"/>-->
-
-          <!-- If TTS is enabled all audio-file params not beginning with -->
-          <!-- '/' 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"/>-->
-          <!-- File to play to acknowledge failure -->
-          <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
-          <!-- File to play to acknowledge muted -->
-          <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
-          <!-- File to play to acknowledge unmuted -->
-          <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
-          <!-- File to play if you are alone in the conference -->
-          <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
-          <!-- File to play when you join the conference -->
-          <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
-          <!-- File to play when you leave the conference -->
-          <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
-          <!-- File to play when you ae ejected from the conference -->
-          <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
-          <!-- File to play when the conference is locked -->
-          <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
-          <!-- File to play to prompt for a pin -->
-          <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
-          <!-- File to play to when the pin is invalid -->
-          <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
-          <!-- Conference pin -->
-          <!--<param name="pin" value="12345"/>-->
-          <!-- Default Caller ID Name for outbound calls -->
-          <param name="caller-id-name" value="FreeSWITCH"/>
-          <!-- Default Caller ID Number for outbound calls -->
-          <param name="caller-id-number" value="8777423583"/>
-        </profile>
+	<profile name="default">
+	  <!-- Sample Rate-->
+	  <param name="rate" value="8000"/>
+	  <!-- Number of milliseconds per frame -->
+	  <param name="interval" value="20"/>
+	  <!-- Energy level required for audio to be sent to the other users -->
+	  <param name="energy-level" value="300"/>
+	  <!-- TTS Engine to use -->
+	  <!--<param name="tts-engine" value="cepstral"/>-->
+	  <!-- TTS Voice to use -->
+	  <!--<param name="tts-voice" value="david"/>-->
+
+	  <!-- If TTS is enabled all audio-file params not beginning with -->
+	  <!-- '/' 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"/>-->
+	  <!-- File to play to acknowledge failure -->
+	  <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
+	  <!-- File to play to acknowledge muted -->
+	  <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
+	  <!-- File to play to acknowledge unmuted -->
+	  <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
+	  <!-- File to play if you are alone in the conference -->
+	  <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
+	  <!-- File to play when you join the conference -->
+	  <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
+	  <!-- File to play when you leave the conference -->
+	  <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
+	  <!-- File to play when you ae ejected from the conference -->
+	  <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
+	  <!-- File to play when the conference is locked -->
+	  <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
+	  <!-- File to play to prompt for a pin -->
+	  <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
+	  <!-- File to play to when the pin is invalid -->
+	  <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
+	  <!-- Conference pin -->
+	  <!--<param name="pin" value="12345"/>-->
+	  <!-- Default Caller ID Name for outbound calls -->
+	  <param name="caller-id-name" value="FreeSWITCH"/>
+	  <!-- Default Caller ID Number for outbound calls -->
+	  <param name="caller-id-number" value="8777423583"/>
+	</profile>
       </profiles>
     </configuration>
 
     <configuration name="enum.conf" description="ENUM Module">
       <settings>
-				<param name="default-root" value="e164.org"/>
+	<param name="default-root" value="e164.org"/>
       </settings>
 
       <routes>
-				<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
-				<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
-				<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
+	<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
+	<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
+	<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
       </routes>
     </configuration>
 
     <configuration name="ivr.conf" description="IVR menus">
-			<menus>
-				<menu name="main" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
-					<entry action="menu-exit" digits="*"/>
-					<entry action="menu-sub" digits="2" param="menu2"/>
-					<entry action="menu-exec-api" digits="3" param="api arg"/>
-					<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
-					<entry action="menu-back" digits="5"/>
-					<entry action="menu-call-transfer" digits="7" param="888"/>
-					<entry action="menu-sub" digits="8" param="menu8"/>>
-				</menu>
-				<menu name="menu8" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
-					<entry action="menu-back" digits="#"/>
-					<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
-					<entry action="menu-top" digits="*"/>
-				</menu>
-				<menu name="menu2" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
-					<entry action="menu-back" digits="#"/>
-					<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
-					<entry action="menu-top" digits="*"/>
-				</menu>
-			</menus>
+      <menus>
+	<menu name="main"
+	      greet-long="/soundfiles/greet-long.wav" 
+	      greet-short="/soundfiles/greet-short.wav"
+	      invalid-sound="/soundfiles/invalid.wav"
+	      exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
+	  <entry action="menu-exit" digits="*"/>
+	  <entry action="menu-sub" digits="2" param="menu2"/>
+	  <entry action="menu-exec-api" digits="3" param="api arg"/>
+	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+	  <entry action="menu-back" digits="5"/>
+	  <entry action="menu-call-transfer" digits="7" param="888"/>
+	  <entry action="menu-sub" digits="8" param="menu8"/>>
+	</menu>
+	<menu name="menu8"
+	      greet-long="/soundfiles/greet-long.wav"
+	      greet-short="/soundfiles/greet-short.wav"
+	      invalid-sound="/soundfiles/invalid.wav"
+	      exit-sound="/soundfiles/exit.wav"
+	      timeout ="15"
+	      max-failures="3">
+	  <entry action="menu-back" digits="#"/>
+	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+	  <entry action="menu-top" digits="*"/>
+	</menu>
+	<menu name="menu2"
+	      greet-long="/soundfiles/greet-long.wav"
+	      greet-short="/soundfiles/greet-short.wav"
+	      invalid-sound="/soundfiles/invalid.wav"
+	      exit-sound="/soundfiles/exit.wav"
+	      timeout ="15"
+	      max-failures="3">
+	  <entry action="menu-back" digits="#"/>
+	  <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+	  <entry action="menu-top" digits="*"/>
+	</menu>
+      </menus>
     </configuration> 
 
   </section>
@@ -462,47 +478,47 @@
     <!-- *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})$">
+	<condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
 	  <action application="enum" data="$1"/>
 	  <action application="bridge" data="${enum_auto_route}"/>
-        </condition>
+	</condition>
       </extension>
 
       <!-- Call the FreeSWITCH conference via SIP -->
-        <!--<extension name="FreeSWITCH Conference SIP">-->
-          <!--<condition field="destination_number" expression="^888$">-->
-            <!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
-          <!--</condition>-->
-        <!--</extension> -->
+      <!--<extension name="FreeSWITCH Conference SIP">-->
+      <!--<condition field="destination_number" expression="^888$">-->
+      <!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
+      <!--</condition>-->
+      <!--</extension> -->
 
       <!-- Call the FreeSWITCH conference via IAX -->
-        <!--<extension name="FreeSWITCH Conference IAX">-->
-          <!--<condition field="destination_number" expression="^8888$">-->
-            <!--<action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>-->
-          <!--</condition>-->
-        <!--</extension>-->
+      <!--<extension name="FreeSWITCH Conference IAX">-->
+      <!--<condition field="destination_number" expression="^8888$">-->
+      <!--<action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>-->
+      <!--</condition>-->
+      <!--</extension>-->
 
       <extension name="testmusic">
-        <condition field="destination_number" expression="^1234$">
+	<condition field="destination_number" expression="^1234$">
 	  <!-- 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="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
-        </condition>
+	  <action application="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
+	</condition>
       </extension>
 
       <!-- Enter an existing conference -->
       <extension name="1000">
-        <condition field="destination_number" expression="^1000$">
-          <action application="conference" data="freeswitch"/>
-        </condition>
+	<condition field="destination_number" expression="^1000$">
+	  <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$">
-          <action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
-        </condition>
+	<condition field="destination_number" expression="^2000$">
+	  <action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
+	</condition>
       </extension>
 
       <!-- extensions starting with 4, all the numbers after 4 form a numeric filename -->
@@ -515,33 +531,33 @@
       <!-- 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>
+	<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>
+	<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/a/a/$1"/>
-        </condition>
+	<condition field="rdnis" expression="8881231234"/>
+	<condition field="destination_number" expression="(.*)">
+	  <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"/>
-        <condition field="destination_number" expression="9999">
-          <action application="playback" data="/var/sounds/beep.gsm"/>
-        </condition>
+	<condition field="source" expression="mod_iax"/>
+	<condition field="destination_number" expression="9999">
+	  <action application="playback" data="/var/sounds/beep.gsm"/>
+	</condition>
       </extension>
 
     </context>