diff --git a/conf/conference.conf.xml b/conf/conference.conf.xml
index a5a3744a14..c5d05dabdb 100644
--- a/conf/conference.conf.xml
+++ b/conf/conference.conf.xml
@@ -1,8 +1,8 @@
 <!-- None of these paths are real if you want any of these options you need to really set them up -->
 <configuration name="conference.conf" description="Audio Conference">
-  <!-- Advertise certian presence on startup . -->
+  <!-- Advertise certain presence on startup . -->
   <advertise>
-    <room name="888@sub.mydomain.com" status="FreeSWITCH"/>
+    <room name="888@$${server_name}" status="FreeSWITCH"/>
   </advertise>
 
   <!-- These are the default keys that map when you do not specify a caller control group -->	
@@ -29,7 +29,7 @@
     <!--If no profile is specified it will default to "default"-->
     <profile name="default">
       <!-- Domain (for presence) -->
-      <param name="domain" value="sub.mydomain.com"/>
+      <param name="domain" value="$${server_name}"/>
       <!-- Sample Rate-->
       <param name="rate" value="8000"/>
       <!-- Number of milliseconds per frame -->
@@ -81,9 +81,9 @@
       <!-- Conference pin -->
       <!--<param name="pin" value="12345"/>-->
       <!-- Default Caller ID Name for outbound calls -->
-      <param name="caller-id-name" value="FreeSWITCH"/>
+      <param name="caller-id-name" value="$${outbound_caller_name}"/>
       <!-- Default Caller ID Number for outbound calls -->
-      <param name="caller-id-number" value="8777423583"/>
+      <param name="caller-id-number" value="$${outbound_caller_id}"/>
     </profile>
   </profiles>
 </configuration>
diff --git a/conf/default_context.xml b/conf/default_context.xml
index 3b450196cd..047d2c3637 100644
--- a/conf/default_context.xml
+++ b/conf/default_context.xml
@@ -29,7 +29,7 @@
   <!-- Call the FreeSWITCH conference via SIP -->
   <!--<extension name="FreeSWITCH Conference SIP">-->
   <!--<condition field="destination_number" expression="^888$">-->
-  <!--<action application="bridge" data="sofia/$${domain}/888@conference.freeswitch.org"/>-->
+  <!--<action application="bridge" data="sofia/$${sip_profile}/888@conference.freeswitch.org"/>-->
   <!--</condition>-->
   <!--</extension> -->
 
@@ -53,7 +53,7 @@
       <!-- 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/$${domain}/1234@conference.freeswitch.org"/>
+      <action application="bridge" data="sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
     </condition>
   </extension>
 
@@ -67,7 +67,7 @@
   <!-- 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/$${domain}/1234@conference.freeswitch.org"/>
+      <action application="conference" data="bridge:mydynaconf:sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
     </condition>
   </extension>
 
diff --git a/conf/dingaling.conf.xml b/conf/dingaling.conf.xml
index de4f699df3..b7b0ae0fc5 100644
--- a/conf/dingaling.conf.xml
+++ b/conf/dingaling.conf.xml
@@ -8,12 +8,13 @@
 
   <!-- Client Profile (Original mode) -->
   <x-profile type="client">
-    <param name="name" value="$${domain}"/>
+    <param name="name" value="$${xmpp_client_profile}"/>
     <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="auto"/>
+    <param name="rtp-ip" value="$${bind_server_ip}"/>
+    <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip"/> -->
     <param name="auto-login" value="true"/>
     <!-- SASL "plain" or "md5" -->
     <param name="sasl" value="plain"/>
@@ -23,9 +24,6 @@
     <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="auto"/> -->
-    <!-- <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 -->
@@ -37,11 +35,10 @@
 
   <!-- Component (Server to Server Login) -->
   <x-profile type="component">
-    <!-- All traffic for *@sub.mydomain.com will come to you -->
-    <param name="name" value="$${subdomain}"/>
+    <param name="name" value="$${xmpp_server_profile}"/>
     <param name="password" value="secret"/>
     <param name="dialplan" value="XML"/>
-    <param name="rtp-ip" value="auto"/>
+    <param name="rtp-ip" value="$${bind_server_ip}"/>
     <param name="server" value="jabber.server.org:5347"/>
     <!-- disable to trade async for more calls -->
     <param name="use-rtp-timer" value="true"/>
diff --git a/conf/enum.conf.xml b/conf/enum.conf.xml
index db77758e7d..3ac7841aee 100644
--- a/conf/enum.conf.xml
+++ b/conf/enum.conf.xml
@@ -4,8 +4,8 @@
   </settings>
 
   <routes>
-    <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/$${domain}/$1"/>
+    <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/$${sip_profile}/$1"/>
     <route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
-    <route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${domain}/$1"/>
+    <route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>
   </routes>
 </configuration>
diff --git a/conf/freeswitch.xml b/conf/freeswitch.xml
index 0d1fdd128b..2e0027a291 100644
--- a/conf/freeswitch.xml
+++ b/conf/freeswitch.xml
@@ -13,34 +13,115 @@
       Don't modify it while freeswitch is running cos it is mem mapped in most cases =D
   -->
 
-  <!--#set "domain=mydomain.com"-->
-  <!--#set "subdomain=sub.mydomain.com"-->
-  <!--#set "default_codecs=PCMU@20i"-->
-  <!--my domain is $${domain}-->
+  <!-- Preprocessor Variables
+       These are introduced when configuration strings must be consistent across modules. 
+   -->
+  <!-- sip_profile 
+       Must be a domain name if you are being a registry server; otherwise
+       can be any string.
+       used by: sofia.conf.xml enum.conf.xml default_context.xml directory.xml
+  --> 
+  <!--#set "sip_profile=mydomain.com"-->
+  <!-- xmpp_client_profile and xmpp_server_profile
+       xmpp_client_profile can be any string. 
+       xmpp_server_profile is appended to "dingaling_" to form the database name
+       containing the "subscriptions" table.
+       used by: dingaling.conf.xml enum.conf.xml 
+  --> 
+  <!--#set "xmpp_client_profile=xmppc"-->
+  <!--#set "xmpp_server_profile=xmpps"-->
+  <!-- bind_server_ip
+       Can be an ip address, a dns name, or "auto". 
+       This determines an ip address available on this host to bind.
+       If you are separating RTP and SIP traffic, you will want to have
+       use different addresses where this variable appears.
+       Used by: sofia.conf.xml dingaling.conf.xml
+  -->
+  <!--#set "bind_server_ip=auto"-->
+  <!-- external_rtp_ip
+       Used as the public IP address for SDP.
+       Can be an ip address or a string like "stun:stun.server.com" 
+       If unspecified, the bind_server_ip value is used.
+       Used by: sofia.conf.xml dingaling.conf.xml
+  -->
+  <!--#set "external_rtp_ip=stun:stun.server.com"-->
+  <!-- server_name
+       A public ip address or DNS name that is used when advertising conference
+       presence or registering sip.
+       Used by: conference.conf.xml
+  --> 
+  <!-- outbound_caller_id and outbound_caller_name
+       The caller ID telephone number we should use when calling out.
+       Used by: conference.conf.xml
+  -->
+  <!--#set "outbound_caller_name=FreeSWITCH"-->
+  <!--#set "outbound_caller_id=8777423583"-->
+
   <section name="configuration" description="Various Configuration">
     <!--#include "switch.conf.xml"-->
     <!--#include "modules.conf.xml"-->
-    <!--#include "spidermonkey.conf.xml"-->
+
+    <!-- Order doesn't matter, but for clarity these are in same order as modules.conf.xml. 
+         If they aren't loaded by modules.conf.xml, then they are ignored.
+    --> 
+    <!-- Loggers -->
+    <!--#include "console.conf.xml"-->
+    <!--#include "syslog.conf.xml"-->
+
+    <!-- Multi-Faceted -->
+    <!--#include "enum.conf.xml"-->
+
+    <!-- XML Interfaces --> 
+    <!--#include "xml_rpc.conf.xml"-->
+    <!--#include "xml_curl.conf.xml"-->
+    <!-- none for mod_xml_cdr -->
+
+    <!-- Event Handlers -->
+    <!--#include "cdr.conf.xml"-->
     <!--#include "event_multicast.conf.xml"-->
     <!--#include "event_socket.conf.xml"-->
+    <!--#include "xmpp_event.conf.xml"-->
+    <!--#include "zeroconf.conf.xml"-->
+
+    <!-- Directory Interfaces -->
+    <!-- none for mod_ldap; dialplan_directory.conf.xml has ldap connection info -->
+
+    <!-- Endpoints -->
+    <!--#include "dingaling.conf.xml"-->
     <!--#include "iax.conf.xml"-->
-    <!--#include "console.conf.xml"-->
-    <!--#include "sofia.conf.xml"-->
-    <!--#include "syslog.conf.xml"-->
-    <!--#include "woomera.conf.xml"-->
-    <!--#include "wanpipe.conf.xml"-->
     <!--#include "portaudio.conf.xml"-->
     <!--#include "alsa.conf.xml"-->
-    <!--#include "zeroconf.conf.xml"-->
-    <!--#include "xmpp_event.conf.xml"-->
-    <!--#include "dialplan_directory.conf.xml"-->
-    <!--#include "dingaling.conf.xml"-->
-    <!--#include "xml_curl.conf.xml"-->
-    <!--#include "xml_rpc.conf.xml"-->
-    <!--#include "rss.conf.xml"-->
+    <!--#include "sofia.conf.xml"-->
+    <!--#include "wanpipe.conf.xml"-->
+    <!--#include "woomera.conf.xml"-->
+
+    <!-- Applications -->
+    <!-- none for mod_bridgecall, mod_commands, mod_echo, mod_park, mod_playback --> 
     <!--#include "conference.conf.xml"-->
-    <!--#include "enum.conf.xml"-->
+    <!-- ivr.conf is used by mod_dptools -->
     <!--#include "ivr.conf.xml"-->
+
+    <!-- Dialplan Interfaces -->
+    <!--#include "dialplan_directory.conf.xml"-->
+    <!-- mod_dialplan_xml is configured in the separate "dialplan" section. -->
+
+    <!-- Codec Interfaces -->
+    <!-- no configuration needed -->
+    <!-- File Format Interfaces -->
+    <!-- no configuration needed -->
+    <!-- Timers -->
+    <!-- no configuration needed -->
+
+    <!-- Languages -->
+    <!--#include "spidermonkey.conf.xml"-->
+    <!-- none for mod_perl -->
+
+    <!-- ASR /TTS -->
+    <!-- none for mod_cepstral -->
+    <!--#include "rss.conf.xml"-->
+
+    <!-- Say -->
+    <!-- none for mod_say_en -->
     <!--#include "mod_cdr.conf.xml"-->
   </section>
   
@@ -48,6 +129,8 @@
     <!--#include "default_context.xml"-->
   </section>
 
+  <!-- mod_dingaling is reliant on the vcard data in the "directory" section. -->
+  <!-- mod_sofia is reliant on the user data for authorization --> 
   <section name="directory" description="User Directory">
     <!--#include "directory.xml"-->
   </section>
@@ -66,3 +149,4 @@
 
 </document>
 
+
diff --git a/conf/sofia.conf.xml b/conf/sofia.conf.xml
index 2b78fba146..afcfd6900c 100644
--- a/conf/sofia.conf.xml
+++ b/conf/sofia.conf.xml
@@ -1,6 +1,6 @@
 <configuration name="sofia.conf" description="sofia Endpoint">
   <profiles>
-    <profile name="$${domain}">
+    <profile name="$${sip_profile}">
       <!-- Outbound Registrations -->
       <gateways>
 	<!--<gateway name="asterlink.com">-->
@@ -28,12 +28,12 @@
 	<param name="sip-port" value="5060"/>
 	<param name="dialplan" value="enum,XML"/>
 	<param name="dtmf-duration" value="100"/>
-	<param name="codec-prefs" value="$${default_codecs}"/>
+	<param name="codec-prefs" value="PCMU@20i"/>
 	<param name="codec-ms" value="20"/>
 	<param name="use-rtp-timer" value="true"/>
 	<param name="rtp-timer-name" value="soft"/>
-	<param name="rtp-ip" value="auto"/>
-	<param name="sip-ip" value="auto"/>
+	<param name="rtp-ip" value="$${bind_server_ip}"/>
+	<param name="sip-ip" value="$${bind_server_ip}"/>
 
 	<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
 	<!--<param name="rtp-rewrite-timestampes" value="true"/>-->
@@ -60,9 +60,8 @@
 	<!-- 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"/> -->
+	<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
+
 	<!-- <param name="ext-sip-ip" value="100.101.102.103"/> -->
 	<!-- VAD choose one (out is a good choice); -->
 	<!-- <param name="vad" value="in"/> -->
diff --git a/scripts/combineconf.pl b/scripts/combineconf.pl
new file mode 100644
index 0000000000..3119d50bb9
--- /dev/null
+++ b/scripts/combineconf.pl
@@ -0,0 +1,49 @@
+#!/usr/bin/perl -w
+use strict;
+
+=head1 NAME
+
+combineconf.pl - expand #include PIs in a freeswitch conf file
+
+=head1 SYNOPSIS
+
+ # cd conf
+ # ../scripts/combineconf.pl freeswitch.xml > freeswitch_combined.xml
+
+=head1 DESCRIPTION
+
+This is recursive, and will take multiple input files on the command line.
+
+You need to run it from the working directory that the relative include paths
+except to be resolved from.
+
+=head1 AUTHOR
+
+Mark D. Anderson (mda@discerning.com)
+Released under same terms as Perl, or alternatively the MPL.
+
+=cut
+
+use IO::File;
+
+sub filter_file {
+    my ($f) = @_;
+    my $fh = $f eq '-' ? \*STDIN : IO::File->new($f, 'r');
+    die "ERROR: Can't open $f: $!\n" unless $fh;
+    while(<$fh>) {
+	if (m/<!--#include\s+"(.*?)"/) {
+	    filter_file($1);
+	} 
+	else {print;}
+    }
+    undef $fh;
+}
+
+sub main {
+    die "Usage: $0 file1 ...\nCombined output goes to stdout. Use '-' as the filename to use stdin." unless @ARGV;
+    for(@ARGV) {
+	filter_file($_);
+    }
+}
+
+main();