Cisco 7811 fw upgrade and call history issue

Hello Everyone,

I’ve been banging my head for 2 weeks now and I can’t seem to find an answer to all of my problems.

We have a bunch Cisco 7811 and all of them is currently registered with my FreePBX. Thanks to all the people who posted here their guides.

Problem #1 is I can’t seem to upgrade its firmware. I’ve tried all available firmware upgrade for third party call manager including the latest release from cisco (25-APR-2017 release). I’m trying to upgrade now from sip78xx.11-0-1-11 to sip78xx.11-0-0MPP-7

I can see on my TFTP logs that phone has successfully downloaded the firmware from TFTP but is unable to upgrade. I am getting this error message (see image 1) and it means that : An internal phone error occurred during the download attempt; reset the phone to correct the issue. Which obviously didn’t solve my problem. I’m attaching my SEP file for everyone’s reference

My second problem is that there’s no record of recent calls (missed call, dialed number, etc) on the phone at all! Is this something with my config or PBX that I am using?

Any help would be greatly appreciated.

<device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>cisco</sshUserId>
    <sshPassword>cisco</sshPassword>
    <devicePool>
    <dateTimeSetting>
    <dateTemplate>D/M/Ya</dateTemplate>
    <timeZone>UTC+08:00 Standard/Daylight Time</timeZone>
    <ntps>
    <ntp>
    <name>10.10.2.115</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>10.10.1.110</processNodeName>
           </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-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>true</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></natAddress>
    <phoneLabel>1002</phoneLabel>
    <stutterMsgWaiting>0</stutterMsgWaiting>
    <callStats>false</callStats>
    <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
    <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
    <startMediaPort>16384</startMediaPort>
    <stopMediaPort>32766</stopMediaPort>
    <sipLines>
    <line button="1">
    <featureID>9</featureID>
    <featureLabel>Kevin Cabrera</featureLabel>
    <proxy>USECALLMANAGER</proxy>
    <port>5060</port>
    <name>1002</name>
    <displayName>Kevin Cabrera</displayName>
    <autoAnswer>
    <autoAnswerEnabled>2</autoAnswerEnabled>
        </autoAnswer>
    <callWaiting>3</callWaiting>
    <authName>1002</authName>
    <authPassword>sfasfsafsa12521521!!!!</authPassword>
    <sharedLine>false</sharedLine>
    <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
    <messagesNumber>3501</messagesNumber>
    <ringSettingIdle>4</ringSettingIdle>
    <ringSettingActive>5</ringSettingActive>
    <contact>1002</contact>
    <forwardCallInfoDisplay>
    <callerName>true</callerName>
    <callerNumber>true</callerNumber>
    <redirectedNumber>false</redirectedNumber>
    <dialedNumber>true</dialedNumber>
        </forwardCallInfoDisplay>
     </line>   
  </sipLines>
    <voipControlPort>5060</voipControlPort>
    <dscpForAudio>184</dscpForAudio>
    <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
    <dialTemplate>dialplan.xml</dialTemplate>
  </sipProfile>
    <commonProfile>
    <phonePassword></phonePassword>
    <backgroundImageAccess>true</backgroundImageAccess>
    <callLogBlfEnabled>1</callLogBlfEnabled>
   </commonProfile>
    <loadInformation>sip78xx.11-0-0MPP-7</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>
    <sshAccess>0</sshAccess>
    <sshPort>22</sshPort>
    <webAccess>0</webAccess>
    <spanToPCPort>1</spanToPCPort>
    <loggingDisplay>1</loggingDisplay>
    <loadServer></loadServer>
   </vendorConfig>
    <versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
    <networkLocale>US</networkLocale>
    <networkLocaleInfo>
    <name>US</name>
    <version>5.0(2)</version>
   </networkLocaleInfo>
    <deviceSecurityMode>1</deviceSecurityMode>
    <authenticationURL></authenticationURL>
    <directoryURL></directoryURL>
    <idleURL></idleURL>
    <informationURL></informationURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <servicesURL></servicesURL>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <transportLayerProtocol>2</transportLayerProtocol>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
    <capf>
    <phonePort>3804</phonePort>
  </capf>
   </capfList>
    <certHash></certHash>
    <encrConfig>false</encrConfig>

</device>

My experience with Cisco phones is that they are VERY particular about their XML config files. While I don’t see anything in there that screams “I’m a problem”, the xmllint program (xml-lint?) is a good resource for a “first stage” check on your phone config files.

Where can I find that?

It’s installed as part of libxml.

I’m pretty sure you log in as ‘root’ and run it.