freeswitch/conf/sip_profiles/internal.xml

<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="internal" domain="$${domain}">
  <!--aliases are other names that will work as a valid profile name for this profile-->
  <aliases>
    <alias name="$${domain}"/>
    <alias name="default"/>
  </aliases>
  <!-- Outbound Registrations -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  </gateways>
  
  <domains>
    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
    <!--<domain name="$${domain}" parse="true"/>-->
    <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
    <!--<domain name="all" alias="true" parse="true"/>-->
  </domains>
  
  <settings>
    <!-- When calls are in no media this will bring them back to media
	 when you press the hold button.
    -->
    <!--<param name="media-option" value="resume-media-on-hold"/> -->
    <!-- This will allow a call after an attended transfer go back to 
	 bypass media after an attended transfer.
    -->
    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="context" value="public"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- port to bind to for sip traffic -->
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${global_codec_prefs}"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <!-- ip address to use for rtp -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <!-- ip address to bind to -->
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <!--<param name="apply-nat-acl" value="rfc1918"/>-->
    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <!--<param name="enable-timer" value="false"/>-->
    <!--<param name="enable-100rel" value="false"/>-->
    <!--<param name="minimum-session-expires" value="120"/>-->
    <param name="apply-inbound-acl" value="domains"/>
    <!--<param name="apply-register-acl" value="domains"/>-->
    <!--<param name="dtmf-type" value="info"/>-->
    <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <!--enable to use presence and mwi -->
    <param name="manage-presence" value="true"/>

    <!-- used to share presence info across sofia profiles -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!-- ************************************************* -->
    
    <!-- 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"/>-->
    <!--<param name="unregister-on-options-fail" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="false"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
    <param name="tls-sip-port" value="5061"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="tlsv1"/>
    
    <!--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"/>-->
    <!--<param name="pass-rfc2833" 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-bypass-media" value="true"/>-->

    <!--Uncomment to set all inbound calls to proxy media mode-->
    <!--<param name="inbound-proxy-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"/> -->

    <!-- accept any authentication without actually checking (not a good feature for most people) -->
    <!-- <param name="accept-blind-auth" value="true"/> -->
    
    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
    <!-- <param name="suppress-cng" 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"/>-->
    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
    <param name="auth-calls" value="true"/>
    <!-- on authed calls, authenticate *all* the packets not just invite -->
    <param name="auth-all-packets" value="false"/>
    <!-- <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"/>
    <param name="rtp-hold-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"/>-->
    <!-- disable register and transfer which may be undesirable in a public switch -->
    <!--<param name="disable-transfer" value="true"/>-->
    <!--<param name="disable-register" value="true"/>-->
    <!--<param name="enable-3pcc" value="true"/>-->
    <!-- use stun when specified (default is true) -->
    <!--<param name="stun-enabled" value="true"/>-->
    <!-- use stun when specified (default is true) -->
    <!-- set to true to have the profile determine stun is not useful and turn it off globally-->
    <!--<param name="stun-auto-disable" value="true"/>-->
    <!-- Choose the realm challenge key. Default is auto_to -->
    <!--<param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/>-->
    <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
  </settings>
</profile>