From c73320807aff2cc10044a17a3c1159b801956b4a Mon Sep 17 00:00:00 2001
From: Brian West <brian@freeswitch.org>
Date: Fri, 9 Nov 2007 18:03:53 +0000
Subject: [PATCH] house keeping.. moving things around a bit more to demo
 various things you can do

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6198 d0543943-73ff-0310-b7d9-9358b9ac24b2
---
 conf/sip_profiles/default.xml | 106 ++++++++++++++++++++++++++++
 conf/sip_profiles/nat.xml     |  19 +++++
 conf/sofia.conf.xml           | 127 +---------------------------------
 conf/vars.xml                 |   1 +
 4 files changed, 127 insertions(+), 126 deletions(-)
 create mode 100644 conf/sip_profiles/default.xml
 create mode 100644 conf/sip_profiles/nat.xml

diff --git a/conf/sip_profiles/default.xml b/conf/sip_profiles/default.xml
new file mode 100644
index 0000000000..5e3565eb62
--- /dev/null
+++ b/conf/sip_profiles/default.xml
@@ -0,0 +1,106 @@
+<profile name="default">
+  <!--aliases are other names that will work as a valid profile name for this profile-->
+  <!-- <aliases> -->
+  <!--  <alias name="default"/> -->
+  <!-- </aliases> -->
+  <!-- Outbound Registrations -->
+  <gateways>
+    <!--<gateway name="asterlink.com">-->
+    <!--/// account username *required* ///-->
+    <!--<param name="username" value="cluecon"/>-->
+    <!--/// auth realm: *optional* same as gateway name, if blank ///-->
+    <!--<param name="realm" value="asterlink.com"/>-->
+    <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
+    <!--<param name="from-domain" value="asterlink.com"/>-->
+    <!--/// account password *required* ///-->
+    <!--<param name="password" value="2007"/>--> 
+    <!--/// replace the INVITE from user with the channel's caller-id ///-->
+    <!--<param name="caller-id-in-from" value="false"/>-->
+    <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
+    <!--<param name="extension" value="cluecon"/>-->
+    <!--/// proxy host: *optional* same as realm, if blank ///-->
+    <!--<param name="proxy" value="asterlink.com"/>-->
+    <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
+    <!--<param name="register-proxy" value="mysbc.com"/>-->
+    <!--/// expire in seconds: *optional* 3600, if blank ///-->
+    <!--<param name="expire-seconds" value="60"/>-->
+    <!--/// do not register ///-->
+    <!--<param name="register" value="false"/>-->
+    <!-- which transport to use for register -->
+    <!--<param name="register-transport" value="udp"/>-->
+    <!--How many seconds before a retry when a failure or timeout occurs -->
+    <!--<param name="retry_seconds" value="30"/>-->
+    <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
+    <!--<param name="caller-id-in-from" value="false"/>-->
+    <!--extra sip params to send in the contact-->
+    <!--<param name="contact-params" value="tport=tcp"/>-->
+    <!--</gateway>-->
+  </gateways>
+  
+  <domains>
+    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
+    <!--<domain name="$${domain}" parse="true"/>-->
+  </domains>
+  
+  <settings>
+    <param name="debug" value="1"/>
+    <param name="rfc2833-pt" value="101"/>
+    <param name="sip-port" value="5062"/>
+    <param name="dialplan" value="XML,enum"/>
+    <param name="dtmf-duration" value="100"/>
+    <param name="codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="use-rtp-timer" value="true"/>
+    <param name="rtp-timer-name" value="soft"/>
+    <param name="rtp-ip" value="$${local_ip_v4}"/>
+    <param name="sip-ip" value="$${local_ip_v4}"/>
+    <!--enable to use presense and mwi -->
+    <param name="manage-presence" value="true"/>
+    <!--max number of open dialogs in proceeding -->
+    <!--<param name="max-proceeding" value="1000"/>-->
+    <!--session timers for all call to expire after the specified seconds -->
+    <!--<param name="session-timeout" value="120"/>-->
+    <!--<param name="multiple-registrations" value="true"/>-->
+    <!--set to 'greedy' if you want your codec list to take precedence -->
+    <param name="inbound-codec-negotiation" value="generous"/>
+    <!-- if you want to send any special bind params of your own -->
+    <!--<param name="bind-params" value="transport=udp"/>-->
+    
+    <!--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"/>-->
+    
+    <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+    
+    <!--Uncomment to set all inbound calls to no media mode-->
+    <!--<param name="inbound-no-media" value="true"/>-->
+    
+    <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
+    <!--<param name="inbound-late-negotiation" value="true"/>-->
+    
+    <!-- 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"/> -->
+    
+    <!--TTL for nonce in sip auth-->
+    <param name="nonce-ttl" value="60"/>
+    <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec 
+	that the originator is using-->
+    <!--<param name="disable-transcoding" 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"/> -->
+    
+    <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
+    <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
+    <!-- rtp inactivity timeout -->
+    <!--<param name="rtp-timeout-sec" value="300"/>-->
+    <!-- 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"/>-->
+    <!--all inbound reg will look in this domain for the users -->
+    <!--<param name="force-register-domain" value="cluecon.com"/>-->
+  </settings>
+</profile>
+
diff --git a/conf/sip_profiles/nat.xml b/conf/sip_profiles/nat.xml
new file mode 100644
index 0000000000..dd8c6ca89d
--- /dev/null
+++ b/conf/sip_profiles/nat.xml
@@ -0,0 +1,19 @@
+<profile name="nat">
+  <settings>
+    <param name="debug" value="1"/>
+    <param name="rfc2833-pt" value="101"/>
+    <param name="sip-port" value="5061"/>
+    <param name="dialplan" value="XML,enum"/>
+    <param name="dtmf-duration" value="100"/>
+    <param name="codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="use-rtp-timer" value="true"/>
+    <param name="rtp-timer-name" value="soft"/>
+    <param name="rtp-ip" value="$${local_ip_v4}"/>
+    <param name="sip-ip" value="$${local_ip_v4}"/>
+    <param name="manage-presence" value="true"/>
+    <param name="inbound-codec-negotiation" value="generous"/>
+    <param name="nonce-ttl" value="60"/>
+    <param name="auth-calls" value="true"/>
+    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
+  </settings>
+</profile>
diff --git a/conf/sofia.conf.xml b/conf/sofia.conf.xml
index 16e4853348..15c1785ba9 100644
--- a/conf/sofia.conf.xml
+++ b/conf/sofia.conf.xml
@@ -1,130 +1,5 @@
 <configuration name="sofia.conf" description="sofia Endpoint">
   <profiles>
-    <profile name="$${sip_profile}">
-      <!--aliases are other names that will work as a valid profile name for this profile-->
-      <aliases>
-	<alias name="default"/>
-      </aliases>
-      <!-- Outbound Registrations -->
-      <gateways>
-	<!--<gateway name="asterlink.com">-->
-	  <!--/// account username *required* ///-->
-	  <!--<param name="username" value="cluecon"/>-->
-	  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
-	  <!--<param name="realm" value="asterlink.com"/>-->
-	  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
-	  <!--<param name="from-domain" value="asterlink.com"/>-->
-	  <!--/// account password *required* ///-->
-	  <!--<param name="password" value="2007"/>--> 
-	  <!--/// replace the INVITE from user with the channel's caller-id ///-->
-	  <!--<param name="caller-id-in-from" value="false"/>-->
-	  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
-	  <!--<param name="extension" value="cluecon"/>-->
-	  <!--/// proxy host: *optional* same as realm, if blank ///-->
-	  <!--<param name="proxy" value="asterlink.com"/>-->
-	  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
-	  <!--<param name="register-proxy" value="mysbc.com"/>-->
-	  <!--/// expire in seconds: *optional* 3600, if blank ///-->
-	  <!--<param name="expire-seconds" value="60"/>-->
-	  <!--/// do not register ///-->
-	  <!--<param name="register" value="false"/>-->
-	  <!-- which transport to use for register -->
-	  <!--<param name="register-transport" value="udp"/>-->
-	  <!--How many seconds before a retry when a failure or timeout occurs -->
-	  <!--<param name="retry_seconds" value="30"/>-->
-	  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
-	  <!--<param name="caller-id-in-from" value="false"/>-->
-	  <!--extra sip params to send in the contact-->
-	  <!--<param name="contact-params" value="tport=tcp"/>-->
-	<!--</gateway>-->
-      </gateways>
-
-      <domains>
-	<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
-	<!--<domain name="$${domain}" parse="true"/>-->
-      </domains>
-
-      <settings>
-	<param name="debug" value="1"/>
-	<param name="rfc2833-pt" value="101"/>
-	<param name="sip-port" value="5060"/>
-	<param name="dialplan" value="XML,enum"/>
-	<param name="dtmf-duration" value="100"/>
-	<param name="codec-prefs" value="$${global_codec_prefs}"/>
-	<param name="use-rtp-timer" value="true"/>
-	<param name="rtp-timer-name" value="soft"/>
-	<param name="rtp-ip" value="$${local_ip_v4}"/>
-	<param name="sip-ip" value="$${local_ip_v4}"/>
-	<!--enable to use presense and mwi -->
-	<param name="manage-presence" value="true"/>
-	<!--max number of open dialogs in proceeding -->
-	<!--<param name="max-proceeding" value="1000"/>-->
-	<!--session timers for all call to expire after the specified seconds -->
-	<!--<param name="session-timeout" value="120"/>-->
-	<!--<param name="multiple-registrations" value="true"/>-->
-	<!--set to 'greedy' if you want your codec list to take precedence -->
-	<param name="inbound-codec-negotiation" value="generous"/>
-	<!-- if you want to send any special bind params of your own -->
-	<!--<param name="bind-params" value="transport=udp"/>-->
-
-	<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
-	<!--<param name="rtp-rewrite-timestampes" value="true"/>-->
-
-	<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
-	<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
-
-	<!--Uncomment to set all inbound calls to no media mode-->
-	<!--<param name="inbound-no-media" value="true"/>-->
-
-	<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
-	<!--<param name="inbound-late-negotiation" value="true"/>-->
-
-	<!-- this lets anything register -->
-	<!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
-	<!-- <param name="accept-blind-reg" value="true"/> -->
-
-	<!--TTL for nonce in sip auth-->
-	<param name="nonce-ttl" value="60"/>
-	<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec 
-	    that the originator is using-->
-	<!--<param name="disable-transcoding" value="true"/>-->
-	<param name="auth-calls" value="true"/>
-	<!-- on authed calls, authenticate *all* the packets not just invite -->
-	<!-- <param name="auth-all-packets" value="true"/> -->
-
-	<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
-	<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
-	<!-- rtp inactivity timeout -->
-	<!--<param name="rtp-timeout-sec" value="300"/>-->
-	<!-- VAD choose one (out is a good choice); -->
-	<!-- <param name="vad" value="in"/> -->
-	<!-- <param name="vad" value="out"/> -->
-	<!-- <param name="vad" value="both"/> -->
-	<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
-	<!--all inbound reg will look in this domain for the users -->
-	<!--<param name="force-register-domain" value="cluecon.com"/>-->
-      </settings>
-    </profile>
-
-    <profile name="nat">
-      <settings>
-	<param name="debug" value="1"/>
-	<param name="rfc2833-pt" value="101"/>
-	<param name="sip-port" value="5061"/>
-	<param name="dialplan" value="XML,enum"/>
-	<param name="dtmf-duration" value="100"/>
-	<param name="codec-prefs" value="$${global_codec_prefs}"/>
-	<param name="use-rtp-timer" value="true"/>
-	<param name="rtp-timer-name" value="soft"/>
-	<param name="rtp-ip" value="$${local_ip_v4}"/>
-	<param name="sip-ip" value="$${local_ip_v4}"/>
-	<param name="manage-presence" value="true"/>
-	<param name="inbound-codec-negotiation" value="generous"/>
-	<param name="nonce-ttl" value="60"/>
-	<param name="auth-calls" value="true"/>
-	<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
-      </settings>
-    </profile>
-
+    <!--#include "sip_profiles/*" -->
   </profiles>
 </configuration>
diff --git a/conf/vars.xml b/conf/vars.xml
index a6e35cfb32..20842bba97 100644
--- a/conf/vars.xml
+++ b/conf/vars.xml
@@ -8,6 +8,7 @@
        used by: sofia.conf.xml enum.conf.xml default_context.xml directory.xml
   --> 
   <!--#set "sip_profile=$${domain}"-->
+  <!--#set "nat_sip_profile=nat_$${domain}"-->
   <!-- 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