7941G with 3rd Party PBX

[IPrick11]

We are attempting to register a Cisco phone to an a 3rd party PBX. I wish this was not the case, however its best effort at this point. My assumptions are that we need to update the cisco phone with sip firmware along with adding a third party sip endpoint license to the PBX. We have SIP firmware on the phones and XML files, but the phone does not attempt to register.

My assumption is we need to edit the XML files but to what extent? Call Server Address, extension, password, etc? Does an XML file need to be customized for each phone?

 

[gnrslash4life]

This is probably a better question for the PBX vendor and how 3rd party SIP devices register.

Certifications:
A+
Network+
CCENT
CCNA Voice
TVOICE
CAPPS

 

[IPrick11]

That is where I started actually, and they pointed me here. Regardless of the PBX, the XML files will need to be edited. Before I have successful registration I would at least like to see attempts to register that get denied. With an Avaya phone I would normally just go into the ADDR menu and change the call server address to the PBX IP Address. But with a Cisco phone that option doesn't exist, nor does the option to log into a web interface of the phone to manipulate settings.

 

[gnrslash4life]

Cisco phones pull their info from CM. They grab registration IP info from option 150 out of DHCP.

Certifications:
A+
Network+
CCENT
CCNA Voice
TVOICE
CAPPS

 

[IPrick11]

Thank you gnrslash4life, is this true for when the phone is running SIP firmware and H.323?

 

[gnrslash4life]

For SIP yes. H323 is not an option. Only SCCP and SIP.

Certifications:
A+
Network+
CCENT
CCNA Voice
TVOICE
CAPPS

 

[ethereal112]

Much like registering Cisco IP Phones to Asterisk OpenSource.

http://www.norgie.net/documentation/cisco_7940g/index.shtml

 

[IPrick11]

We are now able to confirm the phone is getting the XML files but now it just says registering. Wire shark shows its not even attempted to register to the IP address in our XML file (x.x.x.215). Also what is Ethernetport 2000?


<?xml version="1.0"?>

-<Default>


-<callManagerGroup>


-<members>


-<member priority="0">


-<callManager>


-<ports>

<ethernetPhonePort>2000</ethernetPhonePort>


-<mgcpPorts>

<listen>2427</listen>

<keepAlive>2428</keepAlive>

</mgcpPorts>

</ports>

<processNodeName/>

</callManager>

</member>

</members>

</callManagerGroup>

<loadInformation115 model="Cisco 7941G-GE">SIP41.9-4-2SR1-1S</loadInformation115>

<authenticationURL/>

<directoryURL/>

<idleURL/>

<informationURL/>

<messagesURL/>

<servicesURL/>

</Default>









<?xml version="1.0"?>

-<device>

<fullConfig>true</fullConfig>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>admin</sshUserId>

<sshPassword>admin</sshPassword>


-<devicePool>


-<dateTimeSetting>

<dateTemplate>D/M/Y</dateTemplate>

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


-<ntps>


-<ntp>

<name>x.x.x.175</name>

<ntpMode>Unicast</ntpMode>

</ntp>

</ntps>

</dateTimeSetting>


-<callManagerGroup>

<tftpDefault>true</tftpDefault>


-<members>


-<member priority="0">


-<callManager>


-<ports>

<ethernetPhonePort>2000</ethernetPhonePort>

<sipPort>5060</sipPort>

<securedSipPort>5061</securedSipPort>

</ports>

<processNodeName>x.x.x.215</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

</devicePool>


-<commonProfile>

<phonePassword/>

<backgroundImageAccess>true</backgroundImageAccess>

<callLogBlfEnabled>0</callLogBlfEnabled>

</commonProfile>

<loadInformation>SIP41.9-4-2SR1-1S</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>0</webAccess>

<spanToPCPort>1</spanToPCPort>

<loggingDisplay>1</loggingDisplay>

<loadServer/>

</vendorConfig>


-<userLocale>

<name>English_United_States</name>

<uid>1</uid>

<langCode>en_US</langCode>

<version>1.0.0.0-1</version>

<winCharSet>iso-8859-1</winCharSet>

</userLocale>

<networkLocale>United_States</networkLocale>


-<networkLocaleInfo>

<name>United_States</name>

<uid>64</uid>

<version>1.0.0.0-1</version>

</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL/>

<directoryURL/>

<idleTimeout>10</idleTimeout>

<idleURL/>

<informationURL/>

<messagesURL/>

<proxyServerURL/>

<servicesURL/>

<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>

<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>

<dscpForCm2Dvce>96</dscpForCm2Dvce>

<transportLayerProtocol>4</transportLayerProtocol>

<capfAuthMode>0</capfAuthMode>


-<capfList>


-<capf>

<phonePort>2000</phonePort>

</capf>

</capfList>

<certHash/>

<encrConfig>false</encrConfig>


-<sipProfile>


-<sipProxies>

<backupProxy/>

<backupProxyPort/>

<emergencyProxy/>

<emergencyProxyPort/>

<outboundProxy>x.x.x.215</outboundProxy>

<outboundProxyPort>5060</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>true</rfc2543Hold>

<callHoldRingback>2</callHoldRingback>

<localCfwdEnable>true</localCfwdEnable>

<semiAttendedTransfer>true</semiAttendedTransfer>

<anonymousCallBlock>2</anonymousCallBlock>

<callerIdBlocking>0</callerIdBlocking>

<dndControl>0</dndControl>

<remoteCcEnable>true</remoteCcEnable>

</sipCallFeatures>


-<sipStack>

<sipInviteRetx>6</sipInviteRetx>

<sipRetx>10</sipRetx>

<timerInviteExpires>180</timerInviteExpires>

<timerRegisterExpires>120</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>true</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>

<stutterMsgWaiting>1</stutterMsgWaiting>

<callStats>false</callStats>

<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>

<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

<startMediaPort>16384</startMediaPort>

<stopMediaPort>16399</stopMediaPort>

<voipControlPort>5061</voipControlPort>

<dscpForAudio>184</dscpForAudio>

<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>

<dialTemplate>dialplan.xml</dialTemplate>

<phoneLabel>5384</phoneLabel>

<natEnabled>false</natEnabled>

<natAddress/>


-<sipLines>


-<line button="1">

<featureID>9</featureID>

<featureLabel>Line 1</featureLabel>

<name>5384</name>

<displayName>5384</displayName>

<contact>5384</contact>

<proxy>x.x.x.215</proxy>

<port>5060</port>


-<autoAnswer>

<autoAnswerEnabled>2</autoAnswerEnabled>

</autoAnswer>

<callWaiting>3</callWaiting>

<authName>5384@company.com</authName>

<authPassword>53845384</authPassword>

<sharedLine>false</sharedLine>

<messageWaitingLampPolicy>1</messageWaitingLampPolicy>

<messagesNumber>*79</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>