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external can do the same job of the nat profile. I have added an alias to the external profile for nat so it doesn't break backwards compat. 

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9383 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Brian West 2008-08-28 18:25:48 +00:00
parent d78200dc12
commit f9a8bc51bc
3 changed files with 1 additions and 65 deletions
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1
conf/sip_profiles/external.xml
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@ -7,6 +7,7 @@
<aliases>
<alias name="outbound"/>
<alias name="nat"/> <!-- for backwards compatibility -->
</aliases>
<domains>

33
conf/sip_profiles/nat.xml
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@ -1,33 +0,0 @@
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="nat">
<gateways>
<X-PRE-PROCESS cmd="include" data="nat/*.xml"/>
</gateways>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5070"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!--<param name="enable-3pcc" value="true"/>-->
</settings>
</profile>

32
conf/sip_profiles/nat/example.xml
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@ -1,32 +0,0 @@
<include>
<!--<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"/>-->
<!--/// username to use in from: *optional* same as username, if blank ///-->
<!--<param name="from-user" value="cluecon"/>-->
<!--/// 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"/>-->
<!--/// 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>-->
</include>