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204 posts

Master Geek


# 161866 22-Jan-2015 01:16
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Hello fellow GZers,

I'm hoping some kind soul can put me out of my misery!
Helping a NFP with setting up some Cisco 7911G's with 2talk, and I don't seem to be able to get it going properly.

I've installed SIP on the phone (I think)
The XML config for TFTP is below (password removed)

Where am I going wrong (or is everything wrong?!)

<device xsi:type="axl:XIPPhone" ctiid="1566023366">
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-YA</dateTemplate>
<timeZone>UTC Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>sip.2talk.co.nz</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<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>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>Test</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line
button="1">
<featureID>9</featureID>
<featureLabel>Test label</featureLabel>
<proxy>sip.2talk.co.nz</proxy>
<port>5060</port>
<name>1109</name>

<displayName>TestName</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>MYUSERNAME</authName>
<authPassword>MYPASSWORD</authPassword>

<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>700</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>1109</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line
button="2">
<featureID>21</featureID>
<featureLabel>speed dial name goes here</featureLabel>
<speedDialNumber>speed dial actual number goes in here</speedDialNumber>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP11.8-4-1SR1S</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>1</webAccess>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>00:00</displayOnDuration>
<displayIdleTimeout>00:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<networkLocale>New_Zealand</networkLocale>

<networkLocaleInfo>
<name>New_Zealand</name>
<version>5.0(2)</version>
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL>http://www/ipphone/authenticate.php</authenticationURL>
<directoryURL>http://www/ipphone/directory.xml</directoryURL>
<idleURL></idleURL>
<informationURL>http://www/ipphone/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL>proxy:3128</proxyServerURL>
<servicesURL>http://www/ipphone/services.xml</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>

</device>





I would love to change the world, but they won't give me the source code

#BOFH

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876 posts

Ultimate Geek

Lifetime subscriber

  # 1219669 22-Jan-2015 07:11
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Have you looked at the installation guides on 2Talk for the Cisco 7940/60?

The Cisco user guide shows the 7911 and 7940 in the same doc so I guess the 7911 phone config is similar.

http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucme/ip_phones/display_phones/4-3/english/user/guide/ug_display.pdf

Cheers

2Talk Linksys SPA2102 user





Gordy


28260 posts

Uber Geek

Moderator
Trusted
Biddle Corp
Lifetime subscriber

  # 1219679 22-Jan-2015 07:26
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I'll answer this question in a brutally honest way. If you want to be put on out of your misery, buy a decent phone.

These phones were only ever designed for a CUCM environment and the SIP stack was never designed to be used outside that, and has plenty of issues dealing with NAT related issues as they are not designed up be used on different physical subnets to the CUCM server.

Yes they do SIP, and yes they can work, but it doesn't change the fact it's still a bad product to be using for your needs.




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