Due to the complex nature of this scenario and all the REFERing involved sipXbridge is required to be in the signaling path for this to operate correctly.

The following scenarios either work poorly or not at all (there may be more that just don't work as well)

• Attended transfers to media services - Voicemail, Voicemail boxes, conferencing, auto attendant
  • Voicemail - Caller is disconnected
  • Voicemail boxes - Caller appears on hold to user while they leave a voicemail. Call is cleared once caller hangs up
  • Conferencing - Caller is disconnected
  • Auto Attendant - Caller is disconnected
• Blind Transfers to these services work just fine.

<context name="custom">
  <extension name="Park-Call">
    <condition field="destination_number" expression="^(3500)$">
      <action application="answer"/>
      <action application="sleep" data="300"/>
      <action application="valet_park" data="my_lot auto in 3501 3599"/>
    </condition>
  </extension>
  <extension name="Park-Retrieve">
    <condition field="destination_number" expression="^(35\d\d)$">
      <action application="answer"/>
      <action application="valet_park" data="my_lot $1"/>
    </condition>
  </extension>
</context>

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

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

  <settings>
    <param name="debug" value="1"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="9061"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="G722,PCMU@20i,PCMA@20i,speex,L16"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <param name="context" value="custom"/>

    <!-- 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="share_presence"/>-->
    <!--<param name="presence-hosts" value="$org.sipfoundry.sipxconfig.domain.Domain@1"/>-->
    <!--<param name="force-register-domain" value="$org.sipfoundry.sipxconfig.domain.Domain@1"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$org.sipfoundry.sipxconfig.domain.Domain@1"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="scrooge"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="accept-blind-auth" value="true"/>
    <!--
	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-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="enable-3pcc" value="true"/>

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
      <gateways>
        <gateway name="pbx.example.com">
          <param name="proxy" value="pbx.example.com"/>
          <param name="realm" value="pbx.example.com"/>
          <param name="username" value="~~id~media"/>
          <param name="password" value="1kJLRiF9YS"/>
          <param name="register" value="false"/>
        </gateway>
      </gateways>
</profile>

chown -R sipxchange:sipxchange /etc/sipxpbx/freeswitch