Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
New Member

Provision a 7971G sip - with a pbx

Hey

After having some issues I finally got my Cisco VOIP phone 7971G firmware updated to the v.latest

App Load ID is jar70sip.9-2-3TH1-9

I have the device connected direct to a tFTP server, with configuration file only in the tFTP folder. 

Here is a screen shot from the tFTP log:

tftp1.png

*i have deleted the MAC code from the screen shot.

It still shows Unprovisioned on screen. the SEP-MAC-cnf.xml file looks like this:

<device>

  <deviceProtocol>SIP</deviceProtocol>

  <sshUserId>admin</sshUserId>

  <sshPassword>cisco</sshPassword>

  <devicePool>

        <dateTimeSetting>

            <dateTemplate>M/D/Ya</dateTemplate>

            <timeZone>Europe/London</timeZone>

            <ntps>

                <ntp>

                    <name>OUR-PBX-IP-Address</name>

                    <ntpMode>Unicast</ntpMode>

                </ntp>

            </ntps>

        </dateTimeSetting>

     <callManagerGroup>

        <members>

           <member priority="0">

              <callManager>

                 <ports>

                    <ethernetPhonePort>2000</ethernetPhonePort>

                    <sipPort>5060</sipPort>

                    <securedSipPort>5061</securedSipPort>

                 </ports>

                 <processNodeName>OUR-PBX-IP-Address</processNodeName>

              </callManager>

           </member>

        </members>

     </callManagerGroup>

  </devicePool>

  <commonProfile>

     <phonePassword></phonePassword>

     <backgroundImageAccess>true</backgroundImageAccess>

     <callLogBlfEnabled>2</callLogBlfEnabled>

  </commonProfile>

  <loadInformation>SIP70.9-2-3S</loadInformation>

  <vendorConfig>

     <disableSpeaker>false</disableSpeaker>

     <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>

     <pcPort>0</pcPort>

     <settingsAccess>1</settingsAccess>

     <garp>0</garp>

     <voiceVlanAccess>0</voiceVlanAccess>

     <videoCapability>0</videoCapability>

     <autoSelectLineEnable>0</autoSelectLineEnable>

     <webAccess>1</webAccess>

     <spanToPCPort>1</spanToPCPort>

     <loggingDisplay>1</loggingDisplay>

     <loadServer></loadServer>

  </vendorConfig>

  <networkLocale>United_States</networkLocale>

    <networkLocaleInfo>

        <name>United_States</name>

        <uid>64</uid>

        <version>1.0.0.0-1</version>

    </networkLocaleInfo>

  <deviceSecurityMode>1</deviceSecurityMode>

  <authenticationURL>http://OUR-PBX-IP-Address/cisco/services/authentication.php</authenticationURL>

  <directoryURL>http://OUR-PBX-IP-Address/xmlservices/PhoneDirectory.php</directoryURL>

  <idleURL>http://OUR-PBX-IP-Address/xmlservices/index.php</idleURL>

  <informationURL></informationURL>

  <messagesURL></messagesURL>

  <proxyServerURL></proxyServerURL>

  <servicesURL>http://OUR-PBX-IP-Address/xmlservices/index.php</servicesURL>

  <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>

  <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>

  <dscpForCm2Dvce>96</dscpForCm2Dvce>

  <transportLayerProtocol>4</transportLayerProtocol>

  <capfAuthMode>0</capfAuthMode>

  <capfList>

     <capf>

        <phonePort>3804</phonePort>

     </capf>

  </capfList>

  <certHash></certHash>

  <encrConfig>false</encrConfig>

   <sipProfile>

     <sipProxies>

        <backupProxy></backupProxy>

        <backupProxyPort></backupProxyPort>

        <emergencyProxy></emergencyProxy>

        <emergencyProxyPort></emergencyProxyPort>

        <outboundProxy></outboundProxy>

        <outboundProxyPort></outboundProxyPort>

        <registerWithProxy>true</registerWithProxy>

     </sipProxies>

     <sipCallFeatures>

        <cnfJoinEnabled>true</cnfJoinEnabled>

        <callForwardURI>x--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>none</preferredCodec>

     <dtmfAvtPayload>101</dtmfAvtPayload>

     <dtmfDbLevel>3</dtmfDbLevel>

     <dtmfOutofBand>avt</dtmfOutofBand>

     <alwaysUsePrimeLine>false</alwaysUsePrimeLine>

     <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>

     <kpml>3</kpml>

     <natEnabled>0</natEnabled>

     <natAddress></natAddress>

     <stutterMsgWaiting>0</stutterMsgWaiting>

     <callStats>false</callStats>

     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>

     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

     <startMediaPort>16384</startMediaPort>

     <stopMediaPort>32766</stopMediaPort>

     <voipControlPort>5060</voipControlPort>

     <dscpForAudio>184</dscpForAudio>

     <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>

     <dialTemplate>dialplan.xml</dialTemplate>

     <phoneLabel>Office</phoneLabel>

     <sipLines>

        <line button="1">

           <featureID>9</featureID>

           <featureLabel>42</featureLabel>

           <name>42</name>

           <displayName>42</displayName>

           <contact>42</contact>

           <proxy>OUR-PBX-IP-Address</proxy>

           <port>5060</port>

           <autoAnswer>

              <autoAnswerEnabled>2</autoAnswerEnabled>

           </autoAnswer>

           <callWaiting>3</callWaiting>

           <authName>42</authName>

           <authPassword>*removed*</authPassword>

           <sharedLine>false</sharedLine>

           <messageWaitingLampPolicy>1</messageWaitingLampPolicy>

           <messagesNumber>*97</messagesNumber>

           <ringSettingIdle>4</ringSettingIdle>

           <ringSettingActive>5</ringSettingActive>

           <forwardCallInfoDisplay>

              <callerName>true</callerName>

              <callerNumber>false</callerNumber>

              <redirectedNumber>false</redirectedNumber>

              <dialedNumber>true</dialedNumber>

           </forwardCallInfoDisplay>

        </line>

     </sipLines>

  </sipProfile>

</device>

am I missing something?

also.......  currently we use a softphone device on our computers, were in a shared office company (Regus) who block VOIP, we get around this in our softphone by configuring outbound Proxy to proxy.sipoutbound.com:53    where should we add that in this file?

Everyone's tags (4)
718
Views
0
Helpful
0
Replies
CreatePlease to create content