docker/mod/freeswitch/conf/sip_profiles/external.xml

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--
        <alias name="outbound"/>
        <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
    <!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- RFC 5626 : Send reg-id and sip.instance -->
    <!--<param name="enable-rfc-5626" value="true"/> -->
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="none"/>
    <param name="manage-presence" value="false"/>


    <!-- Added for Microsoft Edge browser -->
    <param name="apply-candidate-acl" value="localnet.auto"/>
    <param name="apply-candidate-acl" value="wan_v4.auto"/>
    <param name="apply-candidate-acl" value="rfc1918.auto"/>
    <param name="apply-candidate-acl" value="any_v4.auto"/>

    <!-- used to share presence info across sofia profiles
         manage-presence needs to be set to passive on this profile
         if you want it to behave as if it were the internal profile
         for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <param name="dbname" value="sqlite://memory://file:external?mode=memory&amp;cache=shared"/>
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
    <!--
        DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    -->

    <param name="rtp-ip" value="$${external_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>

    <!--
      Listen only clients somehow run into this timeout
      causing 
        Hangup sofia/external/GLOBAL_AUDIO_76116@10.7.7.1 [CS_EXECUTE] [MEDIA_TIMEOUT]
        [mcs-freeswitch] Dispatching conference new video floor event released 
        [mcs-freeswitch] Received CHANNEL_HANGUP for
    -->
    <param name="rtp-timeout-sec" value="86400"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="enable-3pcc" value="proxy"/>

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
    <param name="tls-only" value="false"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <!--<param name="tls-cert-dir" value=""/>-->
    <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
    <param name="tls-passphrase" value=""/>
    <!-- Verify the date on TLS certificates -->
    <param name="tls-verify-date" value="true"/>
    <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
    <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
    <param name="tls-verify-policy" value="none"/>
    <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
    <param name="tls-verify-depth" value="2"/>
    <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
    <param name="tls-verify-in-subjects" value=""/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>
    <param name="ws-binding"  value="0.0.0.0:5066"/>
    <param name="wss-binding"  value="$${local_ip_v4}:7443"/>
    

    <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
    <param name="rtcp-audio-interval-msec" value="5000"/>
    <param name="rtcp-video-interval-msec" value="5000"/>

    <!-- Cut down in the join time -->
    <param name="dtmf-type" value="info"/>
    <param name="liberal-dtmf" value="true"/>
  </settings>
</profile>