ciscocfg

<?xml version="1.0" encoding="UTF-8"?>
<device>
    <fullConfig>true</fullConfig>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>admin</sshUserId>
    <sshPassword>admin</sshPassword>
    <!--
        After using the above login, you will be prompted again for a username/password.
        Use debug/debug for this second login, and type 'help' for commands.
        (also see note about sshAccess below)
    -->
 <devicePool>
      <dateTimeSetting>
         <dateTemplate>D/M/Ya</dateTemplate>
         <timeZone>GMT Standard/Daylight Time</timeZone>
         <ntps>
              <ntp>
                  <name>us.pool.ntp.org</name>
                  <ntpMode>Unicast</ntpMode>
              </ntp>
         </ntps>
      </dateTimeSetting>
   <callManagerGroup>
         <members>
            <member priority="0">
               <callManager>
                  <ports>
                     <ethernetPhonePort>2000</ethernetPhonePort>
                     <sipPort>5060</sipPort>
                     <securedSipPort>5060</securedSipPort>
                  </ports>
                  <processNodeName>192.168.1.38</processNodeName>
		 <!--
                     'processNodeName' is the fqdn address of your
                        asterisk server.
                    -->
               </callManager>
            </member>
         </members>
     </callManagerGroup>
</devicePool>

<sipProfile>
   <sipProxies>
         <backupProxy></backupProxy>
         <backupProxyPort></backupProxyPort>
         <emergencyProxy></emergencyProxy>
         <emergencyProxyPort></emergencyProxyPort>
         <outboundProxy></outboundProxy>
         <outboundProxyPort></outboundProxyPort>
         <registerWithProxy>true</registerWithProxy>
   </sipProxies>
   <sipCallFeatures>
	<cnfJoinEnabled>true</cnfJoinEnabled>
	<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
	<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
	<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
	<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
	<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
	<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
	<rfc2543Hold>false</rfc2543Hold>
	<callHoldRingback>2</callHoldRingback>
	<localCfwdEnable>true</localCfwdEnable>
	<semiAttendedTransfer>true</semiAttendedTransfer>
	<anonymousCallBlock>2</anonymousCallBlock>
	<callerIdBlocking>2</callerIdBlocking>
	<dndControl>0</dndControl>
	<remoteCcEnable>true</remoteCcEnable>
  </sipCallFeatures>
  <sipStack>
	<sipInviteRetx>6</sipInviteRetx>
	<sipRetx>10</sipRetx>
	<timerInviteExpires>180</timerInviteExpires>
	<timerRegisterExpires>3600</timerRegisterExpires>
	<timerRegisterDelta>5</timerRegisterDelta>
	<timerKeepAliveExpires>120</timerKeepAliveExpires>
	<timerSubscribeExpires>120</timerSubscribeExpires>
	<timerSubscribeDelta>5</timerSubscribeDelta>
	<timerT1>500</timerT1>
	<timerT2>4000</timerT2>
	<maxRedirects>70</maxRedirects>
	<remotePartyID>false</remotePartyID>
	<userInfo>None</userInfo>
  </sipStack>
      <autoAnswerTimer>1</autoAnswerTimer>
      <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
      <autoAnswerOverride>true</autoAnswerOverride>
      <transferOnhookEnabled>false</transferOnhookEnabled>
      <enableVad>false</enableVad>

      <preferredCodec>g711ulaw</preferredCodec>
      <dtmfAvtPayload>101</dtmfAvtPayload>
      <dtmfDbLevel>3</dtmfDbLevel>
      <dtmfOutofBand>avt</dtmfOutofBand>
      <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
      <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
      <kpml>3</kpml>

    <natEnabled>false</natEnabled>
    <natAddress>192.168.1.1</natAddress> <!--I have tried many things here-->

    <phoneLabel>ext:244</phoneLabel> <!--Appears in top right section of phone -->
      <stutterMsgWaiting>1</stutterMsgWaiting> <!-- shutter to ear piece when phone taken off hook -->
      <callStats>false</callStats>
      <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
      <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
      <startMediaPort>6000</startMediaPort>
      <stopMediaPort>40000</stopMediaPort>
     <sipLines>
<!-- Add lines here -->
         <line button="1">
            <featureID>9</featureID>
            <featureLabel>J xxxxxx</featureLabel><!-- Displays next to Line Number -->
 		<name>244</name> <!-- SIP username -->
		<displayName>Jxxxx xxxxxx</displayName> <!-- Name to display on outbound caller ID -->
		<contact>244</contact> <!-- SIP username again -->
            <proxy>192.168.1.38</proxy> <!-- SIP server -->
            <port>5060</port>
            <autoAnswer>
               <autoAnswerEnabled>2</autoAnswerEnabled>

            </autoAnswer>
            <callWaiting>1</callWaiting>
	    <!-- AUTH -->
            <authName>244</authName> <!-- auth SIP username same as <name>-->
            <authPassword>6xxxxxx</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber> <!-- VOICEMAIL number for message button-->
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>

            <forwardCallInfoDisplay>
               <callerName>true</callerName>
               <callerNumber>false</callerNumber>
               <redirectedNumber>false</redirectedNumber>
               <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
	 </line>
	 <line button="2">
       		<featureID>2</featureID>
	       <featureLabel>S xxxxxxx</featureLabel>
       		<speedDialNumber>255</speedDialNumber>
    	</line>
	<line button="3">
       		<featureID>2</featureID>
	       <featureLabel>G xxxxxxe</featureLabel>
       		<speedDialNumber>266</speedDialNumber>
    	</line>
	<line button="4">
       		<featureID>2</featureID>
	       <featureLabel>M xxxxx</featureLabel>
       		<speedDialNumber>277</speedDialNumber>
    	</line>
	<line button="5">
       		<featureID>2</featureID>
	       <featureLabel>M xxxxxxn</featureLabel>
       		<speedDialNumber>288</speedDialNumber>
    	</line>
	<line button="6">
       		<featureID>2</featureID>
	       <featureLabel>M xxxxx</featureLabel>
       		<speedDialNumber>299</speedDialNumber>
    	</line>
    </sipLines>
<!-- Dialplan info goes here -->
      <voipControlPort>5060</voipControlPort>
      <dscpForAudio>184</dscpForAudio>
      <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
      <dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>


<commonProfile>
      <phonePassword></phonePassword>
      <backgroundImageAccess>true</backgroundImageAccess>
      <callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>

<!-- loadInformation forces phone to reset and load Firmware specified if not currently running regardless of what XMLDefault.cnf.xml<strong> says</strong> -->
<loadInformation>SIP41.8-5-4S</loadInformation>

<vendorConfig>
        <disableSpeaker>false</disableSpeaker>
        <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
        <pcPort>0</pcPort>
        <!--
            Enables and disables the Ethernet switch port on the phone so the IP phone can
            have access to an Ethernet connection for a PC connection through the phone.
                0 = Enabled (default).
                1 = Disabled.
        -->

        <settingsAccess>1</settingsAccess>
        <!--
            Enables and disables the Settings button on an IP phone.
                0 = Disabled.
                1 = Enabled (default). The phone user can modify features by using the
                    Settings menu.
                2 = Restricted. The phone user is allowed to access User Preferences and
                    volume settings only.
        -->

        <garp>0</garp>
        <!--
            Enables and disables IP phone response to gratuitous Address Resolution
            Protocol (ARP) messages from the IP phone's Ethernet interface.
                0 = Disabled.
                1 = Enabled (default).
        -->

        <voiceVlanAccess>0</voiceVlanAccess>
        <!--
            Enables and disables spanning, which is the IP phone's access to the voice
            VLAN of the PC to which the IP phone's Ethernet port is connected.
                0 = Enabled (default).
                1 = Disabled.
        -->

        <videoCapability>0</videoCapability>
        <!--
            7975G doesn't have any "real" video capability, it requires (I believe)
            integration with a special pc-only program.  Enabled here just as a
            curiosity for now.
         -->

        <autoSelectLineEnable>0</autoSelectLineEnable>
        <!--
            Enables and disables auto line selection.
                0 = Disabled.
                1 = Enabled (default).
        -->

	<!--
        <sshAccess>0</sshAccess>
	<sshPort>22</sshPort>


            Added as per voip-info instructions for asterisk ssh to work after ver9.2(1)
        -->

	<webAccess>0</webAccess>
        <!--
            Enables and disables web access that allows phone users to configure settings
            and features on User Option web pages.
                0 = Enabled (default).
                1 = Disabled.
                2 = Read Only.  00
        -->

        <daysDisplayNotActive></daysDisplayNotActive> <!-- sunday = 1, sat = 7 -->
        <displayOnTime>08:00</displayOnTime>
        <displayOnDuration>12:00</displayOnDuration>
        <displayIdleTimeout>00:00</displayIdleTimeout>
        <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>

        <loggingDisplay>1</loggingDisplay>
	<loadServer></loadServer>

        <spanToPCPort>1</spanToPCPort>
        <!--
            Enables and disables the path between the Ethernet switch port of an IP phone
            and a connection to a PC.
                0 = Enabled (default).
                1 = Disabled.
        -->
	<!-- COmment out for 7941 and 7961 0=enable
        <sshAccess>0</sshAccess>
        <sshPort>22</sshPort>
        -->

    </vendorConfig>
   <versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<userLocale>
	<name>English_United_States</name>
	<uid>1</uid>
	<langCode>en_US</langCode>
	<version>1.0.0.0-1</version>
	<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
	<networkLocaleInfo>
	<name>United_States</name>
	<uid>64</uid>
	<version>1.0.0.0-1</version>
</networkLocaleInfo>
   <deviceSecurityMode>1</deviceSecurityMode>
   <authenticationURL></authenticationURL>
   <directoryURL>http://192.168.1.38:8088/asterisk/static/directory.xml</directoryURL>  <!--  This is the URL for a CiscoIPPhoneMenu  -->
   <servicesURL>http://192.168.1.38:8088/asterisk/static/directory2.xml</servicesURL>   <idleURL></idleURL>
   <informationURL></informationURL>
   <messagesURL></messagesURL>
   <proxyServerURL></proxyServerURL>
 <!-- QoS -->
 <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
   <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
   <dscpForCm2Dvce>96</dscpForCm2Dvce>
<!-- 4 - default, 2 - prefer UDP, 1 - prefer TCP -->
 <transportLayerProtocol>4</transportLayerProtocol>
   <capfAuthMode>0</capfAuthMode>
   <capfList>
      <capf>
         <phonePort>3804</phonePort>
      </capf>
   </capfList>
   <certHash></certHash>
   <encrConfig>false</encrConfig>
</device>