<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="default">
<!--aliases are other names that will work as a valid profile name for this profile-->
<aliases>
<alias name="$${domain}"/>
</aliases>
<!-- Outbound Registrations -->
<gateways>
<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
</gateways>
<domains>
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
<!--<domain name="$${domain}" parse="true"/>-->
</domains>
<settings>
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<param name="debug" value="0"/>
<param name="sip-trace" value="yes"/>
<!-- <param name="context" value="public"/> -->
<param name="context" value="default"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<!-- <param name="rtp-rewrite-timestamps" value="true"/> -->
<param name="dialplan" value="XML"/>
<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="rtp-ip" value="10.1.16.1"/> -->
<!-- <param name="sip-ip" value="10.1.16.1"/> -->
<param name="hold-music" value="local_stream://moh"/>
<!-- <param name="contact-params" value="transport=tcp"/> -->
<!--enable to use presense and mwi -->
<param name="manage-presence" value="true"/>
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--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-timestamps" 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="false"/>
<!-- 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="1800"/>
<!-- 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>