Trying to get Cisco 7970 working

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randomness

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I have been struggling to get several Cisco 7970G phones working on 3cx. They worked fine on Asterisk running skinny firmware, but although they're pulling the latest SIP firmware and config from TFTP fine, they're just displayed 'Unprovisioned'. I've created the extensions in 3cx and there's no firewall running, but nothing is showing up in the server logs. My SEPxxxxxxxxxxxxxxx.cnf.xml follows - can anyone shed any light?
Code:
<device> 
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId> ** username of 7970 ip phone when logging into ip phone via SSH ** 
<sshPassword>password</sshPassword> ** password of 7970 ip phone when logging into ip phone via SSH ** 
<devicePool> 
<dateTimeSetting> 
<dateTemplate>D-M-YA</dateTemplate> ** Date format ** 
<timeZone>GMT Standard/Daylight Time</timeZone> ** Select your timezone. See this link for more info: http://www.cisco.com/univercd/cc/td/doc ... #wp1071293 ** 
<ntps> 
<ntp> 
<name>uk.pool.ntp.org</name> ** Enter the server name or IP of your NTP Server ** 
<ntpMode>Unicast</ntpMode> 
</ntp> 
</ntps> 
</dateTimeSetting> 
<callManagerGroup> 
<members> 
<member priority="0"> 
<callManager> 
<ports> 
<ethernetPhonePort>2000</ethernetPhonePort> 
<sipPort>5060</sipPort> 
<securedSipPort>5061</securedSipPort> 
</ports> 
<processNodeName>10.10.10.5</processNodeName> ** Enter the name/IP of your 3CX Server ** 
</callManager> 
</member> 
</members> 
</callManagerGroup> 
</devicePool> 
<sipProfile> 
<sipProxies> ** Self explanatory, you can put the 3cx server ip addresses here on port 5060 (optional) ** 
<backupProxy></backupProxy> 
<backupProxyPort></backupProxyPort> 
<emergencyProxy></emergencyProxy> 
<emergencyProxyPort></emergencyProxyPort> 
<outboundProxy></outboundProxy> 
<outboundProxyPort>5060</outboundProxyPort> 
<registerWithProxy>true</registerWithProxy> 
</sipProxies> 
<sipCallFeatures> ** Do not modify any of these settings ** 
<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> ** Do not modify any of these settings ** 
<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>g711ulaw</preferredCodec> ** Select your preferred codec - G711ulaw, G711alaw, or G729a. 
NOTE: All 3 codecs will be sent, but the preferred one will be listed first. But for 3cx put in G711alaw ** 
<dtmfAvtPayload>101</dtmfAvtPayload> 
<dtmfDbLevel>3</dtmfDbLevel> 
<dtmfOutofBand>avt</dtmfOutofBand> 
<alwaysUsePrimeLine>false</alwaysUsePrimeLine> 
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> 
<kpml>3</kpml> 
<natEnabled>0</natEnabled> 
<natAddress></natAddress> 
<phoneLabel>01234567890</phoneLabel> ** Name that will display in upper right hand corner of the phone, its usually best to put your external PSTN or DID number. ** 
<stutterMsgWaiting>2</stutterMsgWaiting> 
<callStats>false</callStats> 
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> 
<startMediaPort>16384</startMediaPort> 
<stopMediaPort>32766</stopMediaPort> 
<sipLines> 
<line button="1"> 
<featureID>9</featureID> ** This must be 9 (this is not for acceessing an outside line) for a regular SIP phone extension. 
Use 21 here to create a speed dial ** 
<featureLabel>10</featureLabel> ** Text that displays next to line button, usually best to put the extension no. of the IP Phone here. ** 
<proxy>10.10.10.5</proxy> ** ip address or server name of your 3cx server** 
<port>5060</port> ** SIP Port 3cx communicates on ** 
<name>Study</name> ** SIP registration name - this is the firstname and last name you create the extension with in 3cx ** 
<displayName>Study</displayName> ** SIP Display name ** 
<autoAnswer> 
<autoAnswerEnabled>2</autoAnswerEnabled> 
</autoAnswer> 
<callWaiting>3</callWaiting> 
<authName>10</authName> ** sip authentication username you set in 3cx ** 
<authPassword>password</authPassword> ** sip authenication password you set in 3cx ** 
<sharedLine>false</sharedLine> 
<messageWaitingLampPolicy>3</messageWaitingLampPolicy> 
<messagesNumber>505</messagesNumber> 
<ringSettingIdle>4</ringSettingIdle> 
<ringSettingActive>5</ringSettingActive> 
<contact></contact> 
<forwardCallInfoDisplay> 
<callerName>true</callerName> 
<callerNumber>false</callerNumber> 
<redirectedNumber>false</redirectedNumber> 
<dialedNumber>true</dialedNumber> 
</forwardCallInfoDisplay> 
</line> ** you can use the extra buttons along the top right hand side of phone as speed dials by using the following below as an example 
<line button="3"> 
<featureID>21</featureID> ** the feature id needs to be set as 21 if its to be used as a speed dial ** 
<featureLabel>Kitchen</featureLabel> ** Label on phone next to speed dial button ** 
<speedDialNumber>11</speedDialNumber> ** the ext or phone number speed dial will dial/ring ** 
</line> 
<line button="4"> 
<featureID>21</featureID> 
<featureLabel>Lounge</featureLabel> 
<speedDialNumber>12</speedDialNumber> 
</line> 
<line button="7"> ** you can configure the speed dial button to dial an external number aswell ** 
<featureID>21</featureID> 
<featureLabel>Bob</featureLabel> 
<speedDialNumber>901234567890</speedDialNumber> 
</line> 
<line button="8"> 
<featureID>21</featureID> 
<featureLabel>Chris</featureLabel> 
<speedDialNumber>901234567890</speedDialNumber> 
</line> 
</sipLines> 

<voipControlPort>5060</voipControlPort> 
<dscpForAudio>184</dscpForAudio> 
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> 
<dialTemplate>dialplan.xml</dialTemplate> ** dial plan set for the phone locally not for 3cx ** 
</sipProfile> 
<commonProfile> 
<phonePassword></phonePassword> 
<backgroundImageAccess>true</backgroundImageAccess> 
<callLogBlfEnabled>2</callLogBlfEnabled> 
</commonProfile> 

<loadInformation>SIP70.8-3-1S</loadInformation> ** SIP load file name needs to be inserted here - refer firware files loaded in tftp for the cisco phone. Do not add ".load"** 
<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> 
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp> 
<networkLocale></networkLocale> 
<networkLocaleInfo> 
<name></name> 
<version>5.0(2)</version> 
</networkLocaleInfo> 
<deviceSecurityMode>0</deviceSecurityMode> 
<authenticationURL></authenticationURL> 
<directoryURL></directoryURL> 
<idleURL></idleURL> 
<informationURL></informationURL> 
<messagesURL></messagesURL> 
<proxyServerURL></proxyServerURL> 
<servicesURL></servicesURL> 
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> 
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> 
<dscpForCm2Dvce>96</dscpForCm2Dvce> a
<transportLayerProtocol>4</transportLayerProtocol> 
<capfAuthMode>0</capfAuthMode> 
<capfList> 
<capf> 
<phonePort>3804</phonePort> 
</capf> 
</capfList> 
<certHash></certHash> 
<encrConfig>false</encrConfig> 
</device>
 
Have you done a search back though the forum? This model has come up before...http://www.3cx.com/forums/help-how-do-i-get-cisco-phones-working-5447.html#p25680
 
leejor said:
Have you done a search back though the forum? This model has come up before...http://www.3cx.com/forums/help-how-do-i-get-cisco-phones-working-5447.html#p25680
I have indeed. That particular post seems to relate just to the firmware, which is loading fine. The problem is that the phones being SIP now don't seem to even try to register.
 
I think this is your problem
Code:
<outboundProxy></outboundProxy> 
<outboundProxyPort>5060</outboundProxyPort> 
<registerWithProxy>true</registerWithProxy>

You have no outbound proxy but said to register with proxy.
 
abc123 said:
I think this is your problem
Code:
<outboundProxy></outboundProxy> 
<outboundProxyPort>5060</outboundProxyPort> 
<registerWithProxy>true</registerWithProxy>

You have no outbound proxy but said to register with proxy.
Ah, good point! Thanks, but have corrected this and still the same (even after a firmware refresh) unfortunately. Even the time and date haven't changed, which suggests to me that the config file isn't being parsed...?
 
Sometimes it happens with the phones.

I find a power off and then back on helps them pick it up again.
 
Tried that, and reflashed the firmware. It comes up with 'Configuring IP' several times, but then reverts to 'Unprovisioned' again :(
 
Anyone got any more thoughts? Still scratching my head... :cry:
 
I can answer for NTP problem. It does not accept FQDN. Please use IP address. Someone in the net as wrongly saying that for SIP, it is able to support both host name and IP. In fact, it only recognize IP address of NTP server

<ntp>
<name>IP address of uk.pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>

For it to work, I am figuring it out myself
 
Just finished read through your SEP.cnf.xml file. Following are the wrong setting that you have

Wrong
<natEnabled>0</natEnabled>

Right
<natEnabled>false</natEnabled>


Wrong
Inside tag <sipLines>, you have another tag <line button="1">. However, you are missing tag <proxy> and <port> inside which is where the 3CX SIP server hostname (or IP) is

Eg.

<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Line 1</featureLabel>
<name>10</name>
<displayName>Name for Line 1</displayName>
<contact>Contact for Line 1</contact>

<proxy>192.168.1.1</proxy>
<port>5060</port>
<authName>10</authName>
<authPassword>10</authPassword>
...
</line>
</sipLines>
 
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