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