freeswitch/conf/sofia.conf.xml

<configuration name="sofia.conf" description="sofia Endpoint">
  <profiles>
    <profile name="$${domain}">
      <!-- Outbound Registrations -->
      <gateways>
	<!--<gateway name="asterlink.com">-->
	  <!--/// account username *required* ///-->
	  <!--<param name="username" value="cluecon"/>-->
	  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
	  <!--<param name="realm" value="asterlink.com"/>-->
	  <!--/// account password *required* ///-->
	  <!--<param name="password" value="2007"/>--> 
	  <!--/// replace the INVITE from user with the channel's caller-id ///-->
	  <!--<param name="caller-id-in-from" value="false"/>-->
	  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
	  <!--<param name="extension" value="cluecon"/>-->
	  <!--/// proxy host: *optional* same as realm, if blank ///-->
	  <!--<param name="proxy" value="asterlink.com"/>-->
	  <!--/// expire in seconds: *optional* 3600, if blank ///-->
	  <!--<param name="expire-seconds" value="60"/>-->
	  <!--/// do not register ///-->
	  <!--<param name="register" value="false"/>-->
	<!--</gateway>-->
      </gateways>
      <settings>
	<param name="debug" value="1"/>
	<param name="rfc2833-pt" value="101"/>
	<param name="sip-port" value="5060"/>
	<param name="dialplan" value="enum,XML"/>
	<param name="dtmf-duration" value="100"/>
	<param name="codec-prefs" value="$${default_codecs}"/>
	<param name="codec-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"/>

	<!--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"/>-->

	<!-- optional ; -->
	<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
	<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
	<!-- <param name="ext-sip-ip" value="100.101.102.103"/> -->
	<!-- 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"/>-->
      </settings>
    </profile>
  </profiles>
</configuration>