CISCO 7965G SIP Firmware unprovisioned! How to fix it...?

[amex.live]

Dear Cisco Support Comunity,

I have a Problem with one Cisco 7965G IP Phone.

I have successed install the SIP Firmware SIP45.9-4-2-1S

I want to use this Phone ONLY with one my SIP Provider. For this reason I have:

>> SIP USER

>> SIP PASSWORD

>> HOSTNAME / IP Address

>> PORT 5060

How can I configure the SEPxxxxxxxxxxxx.cnf.xml with this values to configurate it to work only with this SIP Provider. The network is simply - INET -> Router -> IP Phone.

Can I this SEPxxxxxxxxxxxx.cnf.xml store in the Phone (I do not want every time when I turn the phone on to use TFTP to send it to phone).

When I turn on the phone it "Registerin" and after 5-10 minutes is in mode "Your current option" but the line button is with red X

Network plan is easy...

SIP PROVIDER  --[WAN]-->  MY ROUTER -- [192.168.1.1/24] --> MY 7965G SIP Phone

----------------------------------------------------------------------------------------

>>>>> SEPxxxxxxxxxxxx.cnf.xml

----------------------------------------------------------------------------------------

<?xml version="1.0" encoding="UTF-8"?>

-<device>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>Admin</sshUserId>

<sshPassword>q1w2e3r4t5</sshPassword>

-<devicePool>

-<dateTimeSetting>

<dateTemplate>D.M.YY</dateTemplate>

<timeZone>Central Europe Standard/Daylight Time</timeZone>

-<ntps>

-<ntp>

<name>192.168.1.1</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>77.72.174.129</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

</devicePool>

-<commonProfile>

<phonePassword/>

<backgroundImageAccess>true</backgroundImageAccess>

<callLogBlfEnabled>2</callLogBlfEnabled>

</commonProfile>

<loadInformation>SIP45.9-0-2SR2S</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>1</sshAccess>

<sshPort>22</sshPort>

<webAccess>1</webAccess>

<spanToPCPort>1</spanToPCPort>

<loggingDisplay>1</loggingDisplay>

<loadServer/>

<daysDisplayNotActive/>

<displayOnTime>03:00</displayOnTime>

<displayOnDuration>00:01</displayOnDuration>

<displayIdleTimeout>00:05</displayIdleTimeout>

<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>

</vendorConfig>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL>http://192.168.1.1/ciscoauth.php</authenticationURL>

<directoryURL>http://192.168.1.1/directory.php</directoryURL>

<idleURL/>

<informationURL/>

<messagesURL/>

<proxyServerURL/>

<servicesURL>http://www.arbeitsplatzvernichtung-durch-outsourcing.de/ebtcisco/index.php?userid=987600&timeoffset=1</servicesURL>

<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>

<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>

<dscpForCm2Dvce>96</dscpForCm2Dvce>

<transportLayerProtocol>2</transportLayerProtocol>

<capfAuthMode>0</capfAuthMode>

-<capfList>

-<capf>

<phonePort>3804</phonePort>

</capf>

</capfList>

<certHash/>

<encrConfig>false</encrConfig>

-<sipProfile>

-<sipProxies>

<backupProxy/>

<backupProxyPort/>

<emergencyProxy/>

<emergencyProxyPort/>

<outboundProxy/>

<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>true</natEnabled>

<natAddress>192.168.1.1</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>my phone name</phoneLabel>

-<sipLines>

-<line button="1">

<featureID>9</featureID>

<featureLabel>my label name</featureLabel>

<name>my name</name>

<displayName>my display name</displayName>

<contact></contact>

<proxy/>

<port>5060</port>

-<autoAnswer>

<autoAnswerEnabled>2</autoAnswerEnabled>

</autoAnswer>

<callWaiting>3</callWaiting>

<authName>my sip server address</authName>

<authPassword>my sip password</authPassword>

<sharedLine>false</sharedLine>

<messageWaitingLampPolicy>1</messageWaitingLampPolicy>

<messagesNumber>*97</messagesNumber>

<ringSettingIdle>4</ringSettingIdle>

<ringSettingActive>5</ringSettingActive>

-<forwardCallInfoDisplay>

<callerName>true</callerName>

<callerNumber>true</callerNumber>

<redirectedNumber>false</redirectedNumber>

<dialedNumber>true</dialedNumber>

</forwardCallInfoDisplay>

</line>

-<line button="2">

<featureID>2</featureID>

<featureLabel></featureLabel>

<speedDialNumber></speedDialNumber>

</line>

</sipLines>

</sipProfile>

</device>

----------------------------------------------------------------------------------

>>>> dialplan.xml

----------------------------------------------------------------------------------

<?xml version="1.0"?>

-<DIALTEMPLATE>

<TEMPLATE Timeout="0" MATCH="*#"/>

<TEMPLATE Timeout="4" MATCH="*"/>

</DIALTEMPLATE>

------------------------------------------------------------------------------

Thank you in advanced!

Best regards,

[Leo Laohoo]

1.  Move this thread to Collaboration, Voice and Video >IP Telephony section. 

2.  Post your SEPmacaddress.cnf.xml file.  (Please do not post screen shots.)

3.  Post dialplan.xml file.  (Please do not post screen shots.)

[amex.live]

Any Ideas ? 

[Leo Laohoo]

Any Ideas ? 

1.  Post your SEPmacaddress.cnf.xml file.  (Please do not post screen shots.)

2.  Post dialplan.xml file.  (Please do not post screen shots.)

[amex.live]

Hello,

see my first asking of top of this page. I was upload it but in .txt becouse I can not upload .cnf.xml files. Now I upload You the correct files in .zip

Please check it..

Thank You in Advance!

[Leo Laohoo]

<loadInformation>SIP45.9-0-2SR2S</loadInformation>

Please use 8.5(4).  

<natEnabled>true</natEnabled>
<natAddress>192.168.1.1</natAddress>

If 192.168.1.1 is your router, then change "true" to false and remove NAT address.

Talk to your SIP provider and ask them if they require ALG to be enabled or not.  If they don't need ALG then you need to disable ALG on your router.

[amex.live]

Hello 

Thank You for Your answer.

I load 8.5(4) and I was disable the NAT settings. I load the firmware with TFTP Server. The result is: Unprovisioned 

The Phone Status is (MY PC -> Phone):

- No IPv4 DNS Server

- TFTP Timeout: CTLFile.tlv

- No CTL Installed

- SEPxxxxxxxxxxxx.cnf.xml

- Error Verifying Config Info

When I change the phone Port to my Router the Phone status is (MY Router -> Phone):

- No CLT Installes

- TFTP Error: ram/SEPxxxxxxxxxxxx.cnf.xml

- TFTP Error

I upload You again the SEP and Dialplan files to see it. I use new one. With the old one is the same!

[Leo Laohoo]

I forgot to ask you, you've got a 7965 right?  Flip the phone to the back.  Above the foot-stand, look at the model number of the phone.  I want to know what "version" number is your phone?   Some 7945/7965 have hardware version number 13, 14 or 15.  

[amex.live]

Yes the Phone Model is: CP-7965G

PID VID: CP-7965G  V06

I think that this V06 is the Version number.

More than 1 Week I probe diferent Version of the firmware and diferent configuraiton without result. Yesterday I test new .cnf.xml file and the Phone show the line buttons, The phone want to Register but did not. I see in my Router that the Phone is connected to the Internet but can not register of my SIP Provider. Before one Year I have old Version ot Cisco IP Phone with monochrom display and it work super but this I can not make the right configuration. I do not why... I use my SIP Account with My Iphone with Zoiper and work.

I attach hier the new .cnf.xml file with it the Phone want to Register but did not. I use it with NAT and without NAT the result is the same.

[Leo Laohoo]

Ok, that version should be fine. 

Can you talk to your voice service provider and get some of their details?  What information and setting can they support?

Get a softphone and configure the softphone for your provider.  If the softphone work then you've got a configuration problem with the phone.  If not, then you've got a configuration problem with your voice service provider.

[amex.live]

Hello again Leo Laohoo,

now I have time and test the phone again with other sep configuration.

Now write me "Select your current options" but the phone line Icon is with red X

Now have signal in the phone and when I dialed a phone number after 10 seconds write me Reorder...

I test it with and without NAT - without success...

Any ideas?

The SIP Account work with 5-6 applications great. Before one year I use it with old cisco phone and work again great. I think the problem is with the config file... but I do not know where can I change the settings.

[amex.live]

Hello again Leo Laohoo,

now I have time and test the phone again with other sep configuration.

Now write me "Select your current options" but the phone line Icon is with red X

Now have signal in the phone and when I dialed a phone number after 10 seconds write me Reorder...

I test it with and without NAT - without success...

Any ideas?

The SIP Account work with 5-6 applications great. Before one year I use it with old cisco phone and work again great. I think the problem is with the config file... but I do not know where can I change the settings.

[amex.live]

Hello again Leo Laohoo,

now I have time and test the phone again with other sep configuration.

Now write me "Select your current options" but the phone line Icon is with red X

Now have signal in the phone and when I dialed a phone number after 10 seconds write me Reorder...

I test it with and without NAT - without success...

Any ideas?

The SIP Account work with 5-6 applications great. Before one year I use it with old cisco phone and work again great. I think the problem is with the config file... but I do not know where can I change the settings.

[amex.live]

Hello again Leo Laohoo,

now I have time and test the phone again with other sep configuration.

Now write me "Select your current options" but the phone line Icon is with red X

Now have signal in the phone and when I dialed a phone number after 10 seconds write me Reorder...

I test it with and without NAT - without success...

Any ideas?

The SIP Account work with 5-6 applications great. Before one year I use it with old cisco phone and work again great. I think the problem is with the config file... but I do not know where can I change the settings.

[grml4d001]

first : your ntp serveur is in lan so you have to select broadcast , unicast is for web

second thing is to type the magic formula : usecallmanager  in lines settings...

it is very poisonous to find right order for right parameters with each firmware , have a look at my topic : https://supportforums.cisco.com/discussion/12425111/help-dumb-set-cp-7945g

that should help