freeswitch/conf/vanilla/sip_profiles/internal-ipv6.xml

<profile name="internal-ipv6">
  <!--
      This is an example of a sofia profile setup to listen on IPv6.
  -->
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!--aliases are other names that will work as a valid profile name for this profile-->
  <settings>
    <!-- Can be set to "_undef_" to remove the User-Agent header -->
    <!-- <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="$${internal_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-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_v6}"/>
    <!-- ip address to bind to -->
    <param name="sip-ip" value="$${local_ip_v6}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <!--<param name="enable-100rel" value="false"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <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="$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <!--enable to use presence 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"/>-->
    <!--max number of receiving requests per second (Default: 1000, 0 - unlimited) -->
    <!--<param name="max-recv-requests-per-second" value="0"/>-->
    <!--session timers for all call to expire after the specified seconds -->
    <!--<param name="session-timeout" value="1800"/>-->
    <!--<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="$${internal_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${internal_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

    <!--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"/>-->

    <!-- Let calls hit the dialplan before selecting codec for the a-leg -->
    <param name="inbound-late-negotiation" value="true"/>

    <!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
    <param name="inbound-zrtp-passthru" 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="$${internal_auth_calls}"/>
    <!-- on authed calls, authenticate *all* the packets not just invite -->
    <param name="auth-all-packets" value="false"/>
    <!-- Shouldn't set these on IPv6 -->
    <!-- <param name="ext-rtp-ip" value="$${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"/>-->
    <!--
        These are enabled to make the default config work better out of the box.
        If you need more than ONE domain you'll need to not use these options.

    -->
    <!--all inbound reg will look in this domain for the users -->
    <param name="force-register-domain" value="$${domain}"/>
    <!--all inbound reg will stored in the db using this domain -->
    <param name="force-register-db-domain" value="$${domain}"/>
    <!-- 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"/>-->

    <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
    <!--<param name="disable-srv" value="false" />-->
    <!--<param name="disable-naptr" value="false" />-->

  </settings>
</profile>