FreeSWITCH dialplan -- /conf/dialplan/default.xml

<?xml version="1.0" encoding="utf-8"?>
<!--
    NOTICE:

    This context is usually accessed via authenticated callers on the sip profile on port 5060
    or transfered callers from the public context which arrived via the sip profile on port 5080.

    Authenticated users will use the user_context variable on the user to determine what context
    they can access.  You can also add a user in the directory with the cidr= attribute acl.conf.xml
    will build the domains ACL using this value.
-->
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
  <context name="default">

    <!-- dial external numbers (mobile, landlines) via gsmopen gsm01 interface -->
    <extension name="gsmopen">
      <condition field="destination_number" expression="^(\d{10})$">
        <action application="set" data="instant_ringback=false"/>
        <action application="set" data="bridge_early_media=true"/>
        <action application="set" data="ignore_early_media=false"/>
        <action application="bridge" data="{instant_ringback=false,bridge_early_media=true,ignore_early_media=false}gsmopen/gsm01/+1$1"/>
      </condition>
    </extension>

    <!--
     This extension allows calling any digits of number
     freeswitch will call plivo outbound server on every incoming call
    -->
    <extension name="plivo">
        <condition field="destination_number" expression="^(\d+)$">
<!--        <condition field="destination_number" expression="^5000$"> -->
            <action application="enable_heartbeat" data="60"/>
            <action application="set" data="instant_ringback=false"/>
            <action application="set" data="bridge_early_media=true"/>
            <action application="set" data="ignore_early_media=false"/>
            <action application="socket" data="127.0.0.1:8084 async full"/>
        <!-- increase the call/recording volume -->
        <action application="set_audio_level" data="read 4"/>
        <action application="set_audio_level" data="write 4"/>
        </condition>
    </extension>

    <!--
     This extension allows calling sip:someone@example.com
     freeswitch will call plivo outbound server on every incoming call
    -->
    <extension name="plivo_sip_uri">
        <condition field="destination_number" expression="^sip:(.*)$">
            <action application="enable_heartbeat" data="60"/>
            <action application="set" data="instant_ringback=false"/>
            <action application="set" data="bridge_early_media=true"/>
            <action application="set" data="ignore_early_media=false"/>
            <action application="socket" data="127.0.0.1:8084 async full"/>
        </condition>
    </extension>

  </context>
</include>