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

diff --git a/conf/sip_profiles/default.xml b/conf/sip_profiles/default.xml
new file mode 100644 (file)
index 0000000..5e3565e
--- /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 (file)
index 0000000..dd8c6ca
--- /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 16e485334848392dcace9f6723dbbc14a726a106..15c1785ba9fe23f68df4f6fca7540d70acee6486 100644 (file)
--- 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 a6e35cfb32d9850d9e33cfe1bcaf408216aac400..20842bba97bedc7a49c5aa683b201aa397e6a5a0 100644 (file)
--- 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