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Example Offsite phones

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Below are steps to get remote Exts 20xx able to call one another and call on-site telephones.

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In the default directory, register the phones as normal:

freeswitch/conf/directory/default/2001.xml

  <include>
  <user id="2001" mailbox="2001">
  <params>
  <param name="password" value="1234"/>
  <param name="vm-password" value="2001"/>
  </params>
  <variables>
  <variable name="accountcode" value="2001"/>
  <variable name="user_context" value="default"/>
  <variable name="effective_caller_id_name" value="your caller name"/>
  <variable name="effective_caller_id_number" value="2001"/>
  </variables>
  </user>
  </include>

Next, create a new profile. The example is a doublenat profile set to port 5090:

freeswitch/conf/sip_profiles/doublenat.xml

     <profile name="doublenat">
     <gateways>
     <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
     </gateways>
     <settings>
     <param name="debug" value="0"/>
     <param name="sip-trace" value="no"/>
     <param name="rfc2833-pt" value="101"/>
     <param name="sip-port" value="5090"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="public"/>
     <param name="dtmf-duration" value="100"/>
     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
     <param name="use-rtp-timer" value="true"/>
     <param name="hold-music" value="$${hold_music}"/>
     <param name="rtp-timer-name" value="soft"/>
     <param name="manage-presence" value="false"/>
     <param name="aggressive-nat-detection" value="true"/>
     <param name="apply-nat-acl" value="rfc1918"/> 
     <param name="inbound-codec-negotiation" value="generous"/>
     <param name="nonce-ttl" value="60"/>
     <param name="auth-calls" value="false"/>
     <param name="rtp-timeout-sec" value="1800"/>
     <param name="rtp-ip" value="$${local_ip_v4}"/>
     <param name="sip-ip" value="$${local_ip_v4}"/>
     <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
     <param name="ext-sip-ip" value="$${external_sip_ip}"/>
     <param name="rtp-timeout-sec" value="300"/>
     <param name="rtp-hold-timeout-sec" value="1800"/>
     </settings>
     </profile>

Finally, inside the //freeswitch/conf/dialplan/public.xml, you can put the following. You can add other features the same as the default.xml but this will get the phones going.

freeswitch/conf/dialplan/public.xml

      <extension name="public_extensions"> 
     <condition field="destination_number" expression="^(20[01][0-9])$">
     <action application="bridge" data="sofia/doublenat/$1%$${domain} "/> 
     </condition>
     </extension>

See Also​