<?xml version="1.0"?>
<!DOCTYPE model PUBLIC "-//SIPFoundry//sipXconfig//Model specification 2.0//EN" 
  "http://www.sipfoundry.org/sipXconfig/dtd/setting_2_0.dtd" [
  <!ENTITY types SYSTEM "types.xml">
  ]>
<model>
  &types;
  <group name="reg">
    <label>Registration</label>
    <setting name="displayName" advanced="yes">
      <type>
        <string/>
      </type>
      <value/>
      <description>Display name used for local user interface as well as SIP 
        signaling.</description>
    </setting>
    <setting name="address" advanced="yes">
      <label>User Address</label>
      <type></type>
      <value/>
      <description>user (e.g. '1002'), the implied domain will be the registration 
        server.</description>
    </setting>
    <setting name="label">
      <type>
        <string/>
      </type>
      <value/>
      <description>Text label to appear on the display adjacent to the associated line key. If 
        omitted, the label will be derived from the user part of 'Registration 
        Server'</description>
    </setting>
    <setting name="type">
      <type></type>
      <value>private</value>
      <description>private OR shared. If set to private, use standard call signaling. If set to 
        shared, augment call signaling with call state subscriptions and notifications and use 
        access control for outgoing calls. </description>
    </setting>
    <setting name="thirdPartyName">
      <type></type>
      <value/>
      <description>This field must match the Address value of the other registration which 
        makes up the bridged line. </description>
    </setting>
    <setting name="auth.userId" advanced="yes">
      <type>
        <string/>
      </type>
      <value/>
      <description>User ID to be used for authentication challenges for this registration. If 
        non-empty, will override the 'Reg User' parameter entered into the Authentication 
        submenu off of the Settings menu on the phone.</description>
    </setting>
    <setting name="auth.password" advanced="yes">
      <type>
        <string password="yes"/>
      </type>
      <value/>
      <description>Password to be used for authentication challenges for this registration. If 
        non-empty, will override the 'Reg Password' parameter entered into the Authentication 
        submenu off of the Settings menu on the phone. </description>
    </setting>
    <setting name="ringType">
      <type><integer min="1" max="22"/></type>
      <value>2</value>
      <description>The ring type for each registration can be configured. Note: ring type number
        1 is silent ring. </description>
    </setting>
    <setting name="lineKeys">
      <type><integer min="1" max="6"/></type>
      <value></value>
      <description>The number of line keys on the phone to be associated with registration. 
      Only set to maximum number of lines. If not specified, value is 1</description>
    </setting>
    <setting name="callsPerLineKey">
      <type><integer min="1" max="6"/></type>
      <value/>
      <description>The number of calls or conferences which may be active or on hold per
      line key for a specific registration on the phone.</description>
    </setting>
    <setting name="lcs" unless="1.6">
      <type>
        <string/>
      </type>
      <description>Microsoft 2005 LCS server</description>
    </setting>
    <setting name="outboundProxy.address" unless="1.6">
      <label>Outbound Proxy</label>
      <type>
        <string/>
      </type>
      <value/>
      <description>Only required if both the following conditions are true. First, 
        you want to direct all outgoing SIP traffic to a server other than 
        the address you specified for the registration server.  Second, you want the outbound 
        proxy only for this line. All other lines will continue to send traffic to 
        the registrar.  NOTE: If you want to set the outbound proxy for all 
        lines, goto to the phone settings for "Sip Server"</description>
    </setting>
    <setting name="outboundProxy.port" unless="1.6">
      <label>Outbound Proxy Port</label>
      <type>
        <integer min="0" max="65535"/>
      </type>
      <value/>
    </setting>
    <setting name="outboundProxy.transport" unless="1.6">
      <label>Outbound Proxy Transport</label>
      <type refid="transport"/>
      <value>DNSnaptr</value>
    </setting>
    <group name="server">
      <label>Registration server</label>
      <group name="1">
        <label>Primary registration server</label>
        <setting name="address">
          <profileName>server.1.address</profileName>
          <type>
            <!-- dotted-decimal IP address or host name -->
          </type>
          <value/>
          <description>Primary Registration Server. IP address or host name of a SIP server 
            that accepts registrations.</description>
        </setting>
        <setting name="port">
          <profileName>server.1.port</profileName>
          <type></type>
          <value/>
          <description>0, empty, 1 to 65535</description>
        </setting>
        <setting name="transport">
          <profileName>server.1.transport</profileName>
          <type refid="transport"/>
          <value>DNSnaptr</value>
          <description>DNSnaptr, TCPpreferred or UDPonly. If empty or DNSnaptr and if Address 
            is a hostname and Port is 0 or empty, do NAPTR then SRV look-ups to try to discover 
            the transport, ports and servers, as per RFC 3263. If Address is an IP address, or 
            a port is given, then UDP is used. If set to TCPpreferred: TCP is the preferred 
            transport, UDP is used if TCP fails. If set to UDPonly: Only UDP will be 
            used.</description>
        </setting>
        <setting name="expires">
          <profileName>server.1.expires</profileName>
          <type></type>
          <value/>
          <description>positive integer</description>
        </setting>
        <setting name="expires.overlap" unless="1.6">
          <label>Re-registration interval</label>
          <profileName>server.1.expires.overlap</profileName>
          <type><integer min="5" max="65535"/></type>
          <value>60</value>
          <description>positive integer between 5 and 65535.</description>
        </setting>
        <setting name="register">
          <profileName>server.1.register</profileName>
          <type></type>
          <value/>
          <description>0, 1 enable registration</description>
        </setting>
        <setting name="retryTimeOut">
          <profileName>server.1.retryTimeOut</profileName>
          <type>
            <integer/>
          </type>
          <value/>
          <description></description>
        </setting>
        <setting name="retryMaxCount">
          <profileName>server.1.retryMaxCount</profileName>
          <type>
            <integer/>
          </type>
          <value/>
          <description></description>
        </setting>
        <setting name="expires.lineSeize">
          <profileName>server.1.expires.lineSeize</profileName>
          <type>
            <integer min="1"/>
          </type>
          <value/>
          <description></description>
        </setting>
      </group>
      <group name="2">
        <label>Secondary Registration Server</label>
        <setting name="address">
          <profileName>server.2.address</profileName>
          <type>
            <!-- dotted-decimal IP address or host name -->
          </type>
          <value/>
          <description>Secondary Registration Server. IP address or host name of a SIP server 
            that accepts registrations.</description>
        </setting>
        <setting name="port">
          <profileName>server.2.port</profileName>
          <type>
            <integer min="0" max="65535"/>
          </type>
          <value/>
          <description>0, empty, 1 to 65535</description>
        </setting>
        <setting name="transport">
          <profileName>server.2.transport</profileName>
          <type refid="transport"/>
          <value>DNSnaptr</value>
          <description>DNSnaptr, TCPpreferred or UDPonly. If empty or DNSnaptr and if Address 
            is a hostname and Port is 0 or empty, do NAPTR then SRV look-ups to try to discover 
            the transport, ports and servers, as per RFC 3263. If Address is an IP address, or 
            a port is given, then UDP is used. If set to TCPpreferred: TCP is the preferred 
            transport, UDP is used if TCP fails. If set to UDPonly: Only UDP will be 
            used.</description>
        </setting>
        <setting name="expires">
          <profileName>server.2.expires</profileName>
          <type>
            <integer min="1"/>
          </type>
          <value/>
          <description></description>
        </setting>
        <setting name="expires.overlap" unless="1.6">
          <label>Re-registration interval</label>
          <profileName>server.2.expires.overlap</profileName>
          <type><integer min="5" max="65535"/></type>
          <value>60</value>
          <description>positive integer between 5 and 65535.</description>
        </setting>
        <setting name="register">
          <profileName>server.2.register</profileName>
          <type></type>
          <value/>
          <description>0, 1 enable registration</description>
        </setting>
        <setting name="retryTimeOut">
          <profileName>server.2.retryTimeOut</profileName>
          <type></type>
          <value/>
          <description>empty or non-negative integer</description>
        </setting>
        <setting name="retryMaxCount">
          <profileName>server.2.retryMaxCount</profileName>
          <type></type>
          <value/>
          <description>empty or non-negative integer</description>
        </setting>
        <setting name="expires.lineSeize">
          <profileName>server.2.expires.lineSeize</profileName>
          <type></type>
          <value/>
          <description>positive integer</description>
        </setting>
      </group>
    </group>
  </group>
  <group name="call">
    <label>Call Handling</label>
    <group name="autoOffHook">
      <label>Auto Off Hook/Ring Down</label>
      <description>Phone will automatically dial this contact immediately upon lifting handset 
        (or similiar operation with headset). This is useful for lobby phones or as a 
        integration technique when interfacing with legacy PBXes that manage call 
        state.</description>
      <setting name="enabled">
        <type>
          <boolean/>
        </type>
        <value>0</value>
        <description>If set to 1, a call will be automatically placed to the contact specified 
          upon going off hook on this registration. </description>
      </setting>
      <setting name="contact">
        <value/>
        <description>ASCII encoded string containing digits (the user part of a SIP URL) or a 
          string that constitutes a valid SIP URL (6416 or 6416@domain.com)</description>
      </setting>
    </group>
    <group name="serverMissedCall">
      <label>Missed Calls</label>
      <setting name="enabled">
        <type>
          <boolean/>
        </type>
        <value>0</value>
        <description>If set to 0, all missed-call events will increment the counter If set to 
          1, only missed-call events sent by the server will increment the 
          counter.</description>
      </setting>
    </group>
  </group>
  <group name="divert">
    <label>Diversion</label>
    <description>SoundPoint(R) IP contains a flexible call forward/diversion feature for this 
      line. In all cases, a call will only be diverted if a non-empty contact has been 
      configured. </description>
    <setting name="contact">
      <type>
        <!-- ASCII encoded string containing digits (the user part of a SIP URL) or a string that constitutes a valid
	        SIP URL (6416 or 6416@domain.com)-->
      </type>
      <value/>
      <description>The forward-to contact used for all automatic call diversion features unless 
        overridden by a specific contact of a per-call diversion feature. </description>
    </setting>
    <setting name="autoOnSpecificCaller">
      <type>
        <boolean/>
      </type>
      <value>1</value>
      <description>If set to 1, calls may be diverted using the Auto Divert feature of the 
        directory. This is a global flag. </description>
    </setting>
    <setting name="sharedDisabled">
      <type>
        <boolean/>
      </type>
      <value>1</value>
      <description>If set to 1, all diversion features on this line will be disabled if the 
        line is configured as shared. </description>
    </setting>
    <group name="fwd">
      <label>Forwarding</label>
      <setting name="enabled">
        <type>
          <boolean/>
        </type>
        <value>0</value>
        <description>If set to 1, the user will be able to enable universal call forwarding via 
          the soft key menu. </description>
      </setting>
    </group>
    <group name="busy">
      <label>On Busy</label>
      <setting name="enabled">
        <type>
          <boolean/>
        </type>
        <value>1</value>
        <description>If set to 1, calls will be forwarded on busy to the contact specified 
          below.</description>
      </setting>
      <setting name="contact">
        <type></type>
        <value/>
        <description>Forward-to contact for calls forwarded due to busy status, if empty, 
          Contact will be used. ASCII encoded string containing digits (the user part of a SIP 
          URL) or a string that constitutes a valid SIP URL (6416 or 6416@domain.com) 
          </description>
      </setting>
    </group>
    <group name="noanswer">
      <label>On No Answer</label>
      <setting name="enabled">
        <type>
          <boolean/>
        </type>
        <value>1</value>
        <description>If set to 1, calls will be forwarded on no answer to the contact 
          specified. </description>
      </setting>
      <setting name="timeout">
        <type>
          <integer min="1"/>
        </type>
        <value>60</value>
        <description>Time in seconds to allow altering before initiating the 
          diversion.</description>
      </setting>
      <setting name="contact">
        <type></type>
        <value/>
        <description>Forward-to contact used for calls forwarded due to no answer, if empty, 
          Contact will be used. ASCII encoded string containing digits (the user part of a SIP 
          URL) or a string that constitutes a valid SIP URL (6416 or 6416@domain.com) 
          </description>
      </setting>
    </group>
    <group name="dnd">
      <label>On Do Not Disturb</label>
      <setting name="enabled">
        <type>
          <boolean/>
        </type>
        <value>0</value>
        <description>If set to 1, calls will be forwarded on DND to the contact specified 
          below. </description>
      </setting>
      <setting name="contact">
        <type></type>
        <value/>
        <description>Forward-to contact used for calls forwarded due to DND status, if empty 
          Contact will be used. ASCII encoded string containing digits (the user part of a SIP 
          URL) or a string that constitutes a valid SIP URL (6416 or 6416@domain.com) 
          </description>
      </setting>
    </group>
  </group>
  <!-- SIMILIAR SECTION EXISTS IN phone.xml to control dialplans for entire phone -->
  <group name="line-dialplan">
    <label>Dial Plan</label>
    <setting name="impossibleMatchHandling">
      <type>
        <integer min="0" max="2"></integer>
      </type>
      <value>0</value>
      <description>When present, and if no digitmap setting are set, this attribute overrides 
        the global dial plan in Dial Plan settings.</description>
    </setting>
    <setting name="removeEndOfDial">
      <type>
        <boolean/>
      </type>
      <value>1</value>
      <description>When present, and if no digitmap setting are set, this attribute overrides 
        the global dial plan in Dial Plan settings.</description>
    </setting>
    <group name="digitmap">
      <label>Digit Map</label>
      <setting name="digitmap">
        <type>
          <!-- string compatible with the digit map feature of MGCP described in 2.1.5 of RFC 3435; string is
	          
	          
	          
	          limited to 512 bytes and 20 segments; a comma is also allowed; when reached in the digit map, a comma will
	          turn dial tone back on. -->
        </type>
        <value/>
        <description>When present, this attribute overrides the global dial plan defined 
          Phone's Dial Plan settings</description>
      </setting>
      <setting name="digitmap.timeOut">
        <label>Time out</label>
        <type>
          <integer min="1"/>
        </type>
        <value/>
        <description>When present, this attribute overrides the global dial plan defined 
          Phone's Dial Plan settings</description>
      </setting>
      <group name="routing.1">
        <label>Emergency Gateway</label>
        <setting name="address">
          <type>
            <!-- dotted-decimal IP address or host name -->
          </type>
          <value/>
          <description>IP address or host name and port of a SIP server that will be used for 
            routing calls.</description>
        </setting>
        <setting name="port">
          <type>
            <integer min="1" max="65535"/>
          </type>
          <value/>
          <description></description>
        </setting>
        <setting name="emergency.1.value">
          <label>Emergency Numbers</label>
          <type>
            <!-- Comma separated list of entries or single entry representing a or a combination of SIP
	            URL. -->
          </type>
          <value/>
          <description>This represents the URLs that should be watched for emergency routing. 
            When one of these defined URL is detected as being dialed by the user, the call 
            will be automatically directed to the defined emergency server. </description>
        </setting>
      </group>
    </group>
  </group>
  <group name="msg.mwi">
    <label>Messaging</label>
    <description>SoundPoint(R) IP supports message-waiting indication on a per-registration 
      basis. </description>
    <setting name="subscribe">
      <type>
      </type>
      <value/>
      <description>If non-empty, the phone will send a SUBSCRIBE request to this contact after 
        boot-up. ASCII encoded string containing digits (the user part of a SIP URL) or a 
        string that constitutes a valid SIP URL (6416 or 6416@domain.com)</description>
    </setting>
    <setting name="callBackMode">
      <type>
        <enum>
          <option>
            <value>contact</value>
          </option>
          <option>
            <value>registration</value>
          </option>
          <option>
            <value>disabled</value>
          </option>
        </enum>
      </type>
      <value>disabled</value>
      <description>contact, registration or disabled, If set to 'contact', a call will be 
        placed to the contact specified in the callback attribute when the user invokes message 
        retrieval. If set to 'registration', a call will be placed using this registration to 
        the contact registered (the phone will call itself). If set to 'disabled', message 
        retrieval is disabled. </description>
    </setting>
    <setting name="callBack">
      <type>
        <!--ASCII encoded string containing digits (the user part of a SIP URL) or a string that
	        constitutes a valid SIP URL (6416 or 6416@domain.com)-->
      </type>
      <value/>
      <description>Contact to call when retrieving messages for this registration. 
        </description>
    </setting>
  </group>
</model>
