[cisco-voip] Problem with changing 7975g phone to SIP

Fri Feb 3 14:28:30 EST 2023

Hello,

I am trying to convert a 7975g phone to SIP and have it register to my 
PBX (Firebrick FB2700  latest firmware).

I have done a full reset (3491672850*#) and have successfully updated 
the bootloader and firmware to SIP75.9-4-2-1S.

However I am having trouble provisioning the phone and getting it to 
register with my PBX. I can get as far as the phone saying it is 
registering, but I do not see any SIP traffic from the phone. I am using 
a passive lan tap on the rj45 cable from the phone.

I have tried a number of variations of the XMDefault.cnf.xml file. This 
is the current version I am trying.

<Default>
<callManagerGroup>
<members>
<member  priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>10.151.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation437  model="Cisco IP Phone 7975"></loadInformation437>
</Default>

Similarly with the SEP<mac>.cnf.xml file.

[xml]
<device>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>

<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</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>10.151.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

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

<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://10.151.0.1/cisco/services/authentication.php</authenticationURL>
<directoryURL>http://10.151.0.1/xmlservices/PhoneDirectory.php</directoryURL>
<idleURL>http://10.151.0.1/xmlservices/index.php</idleURL>
<informationURL></informationURL>

<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://10.151.0.1/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>false</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>Roger</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>SipUser</featureLabel>
<name>SipUser</name>
<displayName>SipUser</displayName>
<contact>SipUser</contact>

<proxy>10.151.0.1</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>SipUser</authName>
<authPassword>SipPass</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>
[/xml]

This combination gets the phone into registering state. But no sip 
traffic goes out on the LAN. In common with most attempts it also 
results in the loss of the web server access to the phone.

$ nmap 10.151.0.129
Starting Nmap 7.80 (https://nmap.org  ) at 2023-02-03 19:21 GMT
Nmap scan report for 10.151.0.129
Host is up (0.0011s latency).
All 1000 scanned ports on 10.151.0.129 are closed

Nmap done: 1 IP address (1 host up) scanned in 1.54 seconds

So I have something wrong somewhere, but I cannot figure out what.

Anyone got any ideas?

Thanks.

Roger
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