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Beginner

Cisco CP8800 series ip phone access web interface.

Hello,

Recently purchased a Cisco CP-8861 IP phone that was advertised as specifically designed for 3rd party call control. I started up the phone and it automatically went to an expressway sign-in app. I was able to locate the IP address of the phone and tried accessing the web interface by simply typing in the ip address as typically done with other models. While the phone responds to pings it refuses any connection attempt using it's ip address.

I've looked at all the documentation online numerous times and while I've found several others with the same question I haven't found any answers. I'm wondering if it has to do with upgrading the firmware? If so, can it be done without use of CUCM? I see that a TFTP server might help but I want to make sure that's the issue before doing the upgrade.

My current firmware is sip88xx.10-3-1-20

I really hope there is something simple that I'm overlooking!

1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted
Hall of Fame Community Legend

Go HERE.  At the bottom of the page, there is a configuration file for the 9971.  With a few lines of modification the file will be compatible with the 8861.  

Also in the same section there is a description on how to remotely SSH into the phone.

Hope it helps & please don't forget to rate our useful posts.  

View solution in original post

34 REPLIES 34
Highlighted
Hall of Fame Community Legend

Go HERE.  At the bottom of the page, there is a configuration file for the 9971.  With a few lines of modification the file will be compatible with the 8861.  

Also in the same section there is a description on how to remotely SSH into the phone.

Hope it helps & please don't forget to rate our useful posts.  

View solution in original post

Highlighted

Thank you, this helped out a lot! The phone is now alive! The issue is that it's still not registering with Asterisk. I can call the extension (1006) and it instantly goes to voicemail. There are also a few errors being reported on the phones Web UI. Any idea of what's going on?

Phone reported errors

  • [2:42:34pm 29/03/16] 802.1X Authentication: Timed out
  • [3:07:18pm 29/03/16] No trust list installed
  • [3:07:20pm 29/03/16] SEP[MYMAC].cnf.xml(TFTP)
  • [3:07:23pm 29/03/16] VPN not configured
  • [3:07:25pm 29/03/16] Time zone data download failed
  • [3:07:27pm 29/03/16] Error updating locale
  • [3:07:28pm 29/03/16] Error updating locale
  • [3:07:29pm 29/03/16] TFTP error : dialplan.xml
  • [3:07:31pm 29/03/16] File not found : softKey9971.xml
  • [3:08:15pm 29/03/16] 802.1X Authentication: Timed out

Template File with mod's

<?xml version="1.0" encoding="UTF-8"?>
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<tzdata>
<tzolsonversion>2015a</tzolsonversion>
<tzupdater>tzupdater.jar</tzupdater>
</tzdata>
<devicePool>
<dateTimeSetting>
<name>CMLocal</name>
<dateTemplate>D/M/YYa</dateTemplate>
<timeZone>Central Standard/Daylight Time</timeZone>
<olsonTimeZone>America/Chicago</olsonTimeZone>
<ntps>
<ntp>
<name>time.nist.gov</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>MYSERVERIP</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>sip9971.9-4-2SR1-2</loadInformation>
<featurePolicyFile>DefaultFP.xml</featurePolicyFile>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<ciscoCamera>1</ciscoCamera>
<videoCapability>1</videoCapability>
<usbClasses>0,1,2</usbClasses>
<sdio>1</sdio>
<wifi>0</wifi>
<bluetoothProfile>0,1</bluetoothProfile>
<powerNegotiation>0</powerNegotiation>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>
<g722CodecSupport>2</g722CodecSupport>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>07:00</displayOnTime>
<displayOnDuration>12:00</displayOnDuration>
<displayIdleTimeout>00:15</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<commonConfig>
<usb1>1</usb1>
<usb2>1</usb2>
<ciscoCamera>1</ciscoCamera>
<usbClasses>0,1,2</usbClasses>
<sdio>1</sdio>
<bluetooth>1</bluetooth>
<wifi>0</wifi>
<bluetoothProfile>0,1</bluetoothProfile>
<joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
</commonConfig>
<enterpriseConfig>
<usb1>1</usb1>
<usb2>1</usb2>
<ciscoCamera>1</ciscoCamera>
<usbClasses>0,1,2</usbClasses>
<sdio>1</sdio>
<bluetooth>1</bluetooth>
<wifi>0</wifi>
<bluetoothProfile>0,1</bluetoothProfile>
<joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
<videoCapability>0</videoCapability>
<webAccess>0</webAccess>
<eapAuthentication>2</eapAuthentication>
<webProtocol>0</webProtocol>
</enterpriseConfig>
<advertiseG722Codec>1</advertiseG722Codec>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesNumber></messagesNumber>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://cisco.internect.net/</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<phonePersonalization>1</phonePersonalization>
<rollover>0</rollover>
<singleButtonBarge>0</singleButtonBarge>
<joinAcrossLines>1</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>false</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>1</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>true</retainForwardInformation>
</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>0</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>true</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>10000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
<dscpForTelepresence>128</dscpForTelepresence>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<softKeyFile>softKey9971.xml</softKeyFile>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>Bryan Office</phoneLabel>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>1006</featureLabel>
<name>1006</name>
<displayName>DISPLAY NAME</displayName>
<contact>CONTACT</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>1006</authName>
<authPassword>MYUSERSECRETPHRASE</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>VOICE MAIL NUMBER</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>4</maxNumCalls>
<busyTrigger>2</busyTrigger>
</line>
<line button="2">
<featureID>21</featureID>
<featureLabel>LABEL BUTTON 2</featureLabel>
<speedDialNumber>NUMBER</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="3">
<featureID>21</featureID>
<featureLabel>LABEL BUTTON 3</featureLabel>
<speedDialNumber>NUMBER</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="4">
<featureID>21</featureID>
<featureLabel>LABEL BUTTON 4</featureLabel>
<speedDialNumber>NUMBER</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>
<line button="5">
<featureID>21</featureID>
<featureLabel>LABEL BUTTON 5</featureLabel>
<speedDialNumber>NUMBER</speedDialNumber>
<featureOptionMask>1</featureOptionMask>
</line>

</sipLines>
</sipProfile>
<phoneServices>
<provisioning>0</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="0" category="0">
<name>WORKED Services</name>
<url>http://cisco.internect.net/</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>

Highlighted
Hall of Fame Community Legend

Remove the following lines:  

<tzdata>
<tzolsonversion>2015a</tzolsonversion>
<tzupdater>tzupdater.jar</tzupdater>
</tzdata>
<softKeyFile>softKey9971.xml</softKeyFile>

[3:07:29pm 29/03/16] TFTP error : dialplan.xml

Do you have a dialplan.xml file?

 

Highlighted

I deleted that and made sure my dialplan.xml file was uploaded this time. Unfortunately still no joy. My dial plan seems pretty simplistic, that wouldn't cause the phone not to answer would it?

Phone Logs

[9:04:16am 30/03/16] No trust list installed
[9:04:18am 30/03/16] SEP[MAC].cnf.xml(TFTP)
[9:04:21am 30/03/16] VPN not configured
[9:04:23am 30/03/16] Error updating locale
[9:04:24am 30/03/16] Error updating locale
[9:05:13am 30/03/16] 802.1X Authentication: Timed out

dialplan.xml

<DIALTEMPLATE>
<TEMPLATE MATCH="5.." TIMEOUT="0"/><!-- Internal Extensions 500 to 599. Dial immediately. -->
<TEMPLATE MATCH="9,1.........." TIMEOUT="0" Tone="Bellcore-Alerting"/><!-- 9+1+10-digits. Dial immediately. -->
<TEMPLATE MATCH="9,.........." TIMEOUT="0"/><!-- Anything else. -->
</DIALTEMPLATE>

Highlighted
Hall of Fame Community Legend

The issue is that it's still not registering with Asterisk. I can call the extension (1006) and it instantly goes to voicemail.

What is not "registering with Asterisk"?  The phone can't register to Asterisk or the Asterisk can't register with the voice service provider?

Highlighted

We are hosting Asterisk onsite so getting the voicemail does mean that there isn't any issues with our SIP trunk. I have the credentials setup correctly in both Asterisk and the phone's cnf.xml file. Is there something basic that I'm forgetting to do?

Highlighted
Hall of Fame Community Legend

We are hosting Asterisk onsite so getting the voicemail does mean that there isn't any issues with our SIP trunk.

That's incorrect.  Asterisk has VM enabled by default.

I have the credentials setup correctly in both Asterisk and the phone's cnf.xml file.

 There are three "parts" of an Asterisk setup. 

1.  IP phone to Asterisk registration; 

2.  Asterisk to voice service provider registration; and 

3.  Asterisk system settings. 

Now, which bit of the three is/are working and which one ain't.  

The issue is that it's still not registering with Asterisk. I can call the extension (1006) and it instantly goes to voicemail.

Excuse me for my coffee-deprived sanity, but this tells me the phone isn't registered to Asterisk and, when using a different phone, any calls to the phone goes to Asterisk voicemail. 

If this is the case, then part 1 isn't setup correctly and part 2 could also not be setup correctly.

Highlighted

Thanks, I fully agree. The point that it's reaching the voicemail helps me know that it's at least getting to Asterisk. I do believe the issue is the phone isn't registering with Asterisk. If I didn't have these phones given to me I would have definitely gone with the SPA series as I never had an issue with them.

Thanks for all your help, hopefully there will be more native support for the 8800 series in the future.

If you have any other suggestions please pass my way.

Highlighted

Hmm... I think I may have found the issue. The firmware I have is the sip version but not the 3rd party control version. I didn't know there were two different SIP versions. I will have to try uploading the other firmware and see if that makes a difference.

Highlighted
Hall of Fame Community Legend

If you have any other suggestions please pass my way.

I can help troubleshoot but I need more information.  

Highlighted

I just tried updating the firmware to the 3rd party sip version but I got an error during upload. Do you know what FailureReason=3 means?

[2:56:35pm 30/03/16] DeviceImageDownloadFailure DeviceName=SEP[Mac] IPv4Address=192.168.1.107 IPv6Address= Method=3 FailureReason=3 Active=sip88xx.10-3-1-20 Inactive=cert.os.mfg.drop FailedLoadId=sip88xx.10-3-1-9-3PCC Server=192.168.1.108

Highlighted
Hall of Fame Community Legend

The firmware I have is the sip version but not the 3rd party control version. I didn't know there were two different SIP versions.

I have never tried TPCC but my "gut" feeling means it's not meant for this purpose you're trying to accomplish.  

I don't have an 88XX phone but one of the things I'd be looking at, to determine if the phone has registered with Asterisk, is the status page.  This page will determine the firmware version as well as the known call manager server "detected".  If the phone can't/didn't register with Asterisk then this column would be blank.  

Highlighted

Yea I've had that screen up pretty constantly. 

Firmware is always sip88xx10-3-1-20 and the active server field has always been blank.

When I get back into the office I'm going to fiddle with the fail2ban, I've read that has caused issues in the past.

Highlighted
Hall of Fame Community Legend

active server field has always been blank.

This means the phone hasn't registered to Asterisk. 

fiddle with the fail2ban

Yes, I've been caught with fail2ban.  Check if the phone is in the blacklist with the command "iptables -L".  

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