Cisco IP Phone 7942 SIP - dont connect to server PBX.

[kdelgado]

Good day,

I have a cisco phone with SIP firmware (SIP42.9-4-2SR2-2S), attached configuration file.

But it does not connect to the PBX (Grandstream) server, nor does it generate output traffic to the IP address of the PBX server 10.1.14.2.

With sniffer I do not see any outbound traffic to the PBX server, it just did the TFTP server I used for the firmware configuration.

Can anybody help me please? Do I need to make the phone try to connect to the PBX server?

Thank you very much!

[Leo Laohoo]

Go HERE.

I've already helped someone use a Cisco phone for Grandstream. 

Is the phone getting a valid IP address? 

In the Status menu, what error messages are displayed? 

Does Grandstream support NAT and ALG or not?

[kdelgado]

Hi 

The phone receives a valid IP address on the same network segment of Grandstream (UCM6102)

In the status menu, display the following errors:

- TFTP Error: softkey.xml 

- Error Updating Locale

- No CTL Installed

- File Not Found: CTLFile.tlv

Grandstream support NAT, attachment configuration NAT.

Thanks so much.

[Leo Laohoo]

- TFTP Error: softkey.xml 

Not important for now.

- Error Updating Locale

- No CTL Installed

- File Not Found: CTLFile.tlv

Not important. 

Grandstream support NAT, attachment configuration NAT.

1.  Did you factory-default the phone? 

2.  Can you post the SEPmacaddress.cnf.xml file?

[kdelgado]

Thanks for you response!

Post configuration file.

<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword><!-- SSH PASSWORD -->
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>UTC Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>10.1.14.2</name><!-- NTP SERVER IP/HOSTNAME (I have NTP enabled on my UCM) -->
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>10.1.14.2</processNodeName><!-- SIP ENDPOINT -->
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP42.8-5-4S</loadInformation><!-- Firmware Version eg SIP41.8-4-4S derived from SIP41.8-4-4S.loads -->
<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>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>6.0.1.1(1)</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>6.0.1.1(1)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></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>5060</backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-serviceuri-cfwdall</callForwardURI>
<callPickupURI>pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>gpickup</callPickupGroupURI>
<meetMeServiceURI>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>2</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>
<phoneLabel>CompanyName</phoneLabel><!-- Label to show opposite the clock -->
<sipLines>
<!-- LINE 1 -->
<line button="1">
<featureID>9</featureID>
<featureLabel>Floyd</featureLabel><!-- LINE NAME -->
<proxy>10.1.14.2</proxy><!-- SIP ENDPOINT -UCM IP ADDRESS -->
<port>5060</port>
<name>131</name><!-- SIP USERNAME -->
<displayName>131</displayName><!-- SIP Caller ID -->
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>131</authName><!-- SIP USERNAME -->
<authPassword>iL9g%237sc7yA6r</authPassword><!-- SIP PASSWORD -->
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>131</contact><!-- SIP USERNAME -->
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<!-- LINE 2 IF NEEDED -->
<!--<line button="2">
COPY FROM LINE 1
</line>-->
</sipLines>
<!-- SOFTKEY DEFINITIONS -->
<softKeyFile>softkey.xml</softKeyFile>
<!-- DIALPLAN -->
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
</device>

[Leo Laohoo]

<transportLayerProtocol>4</transportLayerProtocol>

Try using a value of "1".

<natEnabled>false</natEnabled>
<natAddress></natAddress>

 So who is doing the NAT?  Your modem/router or Grandstream?