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<?xml version="1.0" encoding="utf-8"?>
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<!--
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    NOTICE:
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    This context is usually accessed via authenticated callers on the sip profile on port 5060 
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    or transfered callers from the public context which arrived via the sip profile on port 5080.
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    Authenticated users will use the user_context variable on the user to determine what context
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    they can access.  You can also add a user in the directory with the cidr= attribute acl.conf.xml
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    will build the domains ACL using this value.
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-->
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<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
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<include>
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  <context name="default">
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    <!-- dial external numbers (mobile, landlines) via gsmopen gsm01 interface -->
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    <extension name="gsmopen">
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      <condition field="destination_number" expression="^(\d{10})$">
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        <action application="bridge" data="gsmopen/gsm01/+1$1"/>
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        <action application="set" data="instant_ringback=false"/>
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        <action application="set" data="bridge_early_media=true"/>
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        <action application="set" data="ignore_early_media=false"/>
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        <action application="bridge" data="{instant_ringback=false,bridge_early_media=true,ignore_early_media=false}gsmopen/gsm01/+1$1"/>
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      </condition>
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    </extension>
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    <!--
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     This extension allows calling any digits of number
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     freeswitch will call plivo outbound server on every incoming call
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    -->
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    <extension name="plivo">
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        <condition field="destination_number" expression="^(\d+)$">
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<!--        <condition field="destination_number" expression="^5000$"> -->
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            <action application="enable_heartbeat" data="60"/>
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            <action application="set" data="instant_ringback=false"/> 
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            <action application="set" data="bridge_early_media=true"/> 
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            <action application="set" data="ignore_early_media=false"/>
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            <action application="socket" data="127.0.0.1:8084 async full"/>
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	    <!-- increase the call/recording volume -->
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	    <action application="set_audio_level" data="read 4"/>
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	    <action application="set_audio_level" data="write 4"/>
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        </condition>
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    </extension>
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    <!--
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     This extension allows calling sip:[email protected]
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     freeswitch will call plivo outbound server on every incoming call
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    -->
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    <extension name="plivo_sip_uri">
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        <condition field="destination_number" expression="^sip:(.*)$">
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            <action application="enable_heartbeat" data="60"/>
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            <action application="set" data="instant_ringback=false"/> 
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            <action application="set" data="bridge_early_media=true"/> 
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            <action application="set" data="ignore_early_media=false"/>
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            <action application="socket" data="127.0.0.1:8084 async full"/>
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        </condition>
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    </extension>
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  </context>
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</include>