freeswitch/conf/sip_profiles/default.xml

<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"/>-->
    <!--/// 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"/>--> 
    <!--/// 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>