<commit_before><profile name="internal">
  <!--
      This is a sofia sip profile/user agent.  This will service exactly one ip and port.
      In FreeSWITCH you can run multiple sip user agents on their own ip and port.

      When you hear someone say "sofia profile" this is what they are talking about.
  -->

  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!--aliases are other names that will work as a valid profile name for this profile-->
  <aliases>
    <!--
        <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"/>-->
    <domain name="all" alias="true" parse="false"/>
  </domains>

  <settings>


    <!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
    <!-- <param name="rtp-digit-delay" value="40"/>-->

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

    <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
    <!-- <param name="presence-proto-lookup" value="true"/> -->


    <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
    <param name="liberal-dtmf" value="true"/>


    <!--
        Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
        responding. These options allow you to enable and control a watchdog
        on the Sofia SIP stack so that if it stops responding for the
        specified number of milliseconds, it will cause FreeSWITCH to crash
        immediately. This is useful if you run in an HA environment and
        need to ensure automated recovery from such a condition. Note that if
        your server is idle a lot, the watchdog may fire due to not receiving
        any SIP messages. Thus, if you expect your system to be idle, you
        should leave the watchdog disabled. It can be toggled on and off
        through the FreeSWITCH CLI either on an individual profile basis or
        globally for all profiles. So, if you run in an HA environment with a
        master and slave, you should use the CLI to make sure the watchdog is
        only enabled on the master.
        If such crash occurs, FreeSWITCH will dump core if allowed. The
        stacktrace will include function watchdog_triggered_abort().
    -->
    <param name="watchdog-enabled" value="no"/>
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>

    <param name="log-auth-failures" value="false"/>
    <param name="forward-unsolicited-mwi-notify" value="false"/>

    <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="$${global_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="rtp-ip" value="<%= @mncc_ip_address %>"/>
    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="sip-ip" value="<%= @mncc_ip_address %>"/>
    <!--<param name="hold-music" value="$${hold_music}"/>-->
    <param name="apply-nat-acl" value="nat.auto"/>


    <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
    <!-- <param name="cid-in-1xx" value="false"/> -->

    <!-- extended info parsing -->
    <!-- <param name="extended-info-parsing" value="true"/> -->

    <param name="aggressive-nat-detection" value="true"/>
    <!--<commit_msg>I can see no reason to have nat on the internal profile<commit_after><profile name="internal">
  <!--
      This is a sofia sip profile/user agent.  This will service exactly one ip and port.
      In FreeSWITCH you can run multiple sip user agents on their own ip and port.

      When you hear someone say "sofia profile" this is what they are talking about.
  -->

  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!--aliases are other names that will work as a valid profile name for this profile-->
  <aliases>
    <!--
        <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"/>-->
    <domain name="all" alias="true" parse="false"/>
  </domains>

  <settings>


    <!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
    <!-- <param name="rtp-digit-delay" value="40"/>-->

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

    <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
    <!-- <param name="presence-proto-lookup" value="true"/> -->


    <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
    <param name="liberal-dtmf" value="true"/>


    <!--
        Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
        responding. These options allow you to enable and control a watchdog
        on the Sofia SIP stack so that if it stops responding for the
        specified number of milliseconds, it will cause FreeSWITCH to crash
        immediately. This is useful if you run in an HA environment and
        need to ensure automated recovery from such a condition. Note that if
        your server is idle a lot, the watchdog may fire due to not receiving
        any SIP messages. Thus, if you expect your system to be idle, you
        should leave the watchdog disabled. It can be toggled on and off
        through the FreeSWITCH CLI either on an individual profile basis or
        globally for all profiles. So, if you run in an HA environment with a
        master and slave, you should use the CLI to make sure the watchdog is
        only enabled on the master.
        If such crash occurs, FreeSWITCH will dump core if allowed. The
        stacktrace will include function watchdog_triggered_abort().
    -->
    <param name="watchdog-enabled" value="no"/>
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>

    <param name="log-auth-failures" value="false"/>
    <param name="forward-unsolicited-mwi-notify" value="false"/>

    <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="$${global_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="rtp-ip" value="<%= @mncc_ip_address %>"/>
    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
    <param name="sip-ip" value="<%= @mncc_ip_address %>"/>
    <!--<param name="hold-music" value="$${hold_music}"/>-->
    <param name="apply-nat-acl" value="nat.auto"/>


    <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
    <!-- <param name="cid-in-1xx" value="false"/> -->

    <!-- extended info parsing -->
    <!-- <param name="extended-info-parsing" value="true"/> -->

    <param name="aggressive-nat-detection" value="false"/>
    <!--
{'fraction_non_alphanumeric': 0.14108135421930268, 'fraction_numerical': 0.006872157655381505, 'mean_word_length': 2.9176563737133807, 'pattern_counts': {'":': 0, '<': 155, '<?xml version=': 0, '>': 153, 'https://': 0, 'lorem ipsum': 0, 'www.': 0, 'xml': 4}, 'pii_count': 0, 'substrings_counts': 0, 'word_list_counts': {'cursed_substrings.json': 0, 'profanity_word_list.json': 0, 'sexual_word_list.json': 0, 'zh_pornsignals.json': 0}}
{'dir': 'git-commits-cleaned', 'id': '1780272', 'n_tokens_mistral': 2902, 'n_tokens_neox': 2775, 'n_words': 1073}