- Sun 18th Jun 2017, 09:16
#4698
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Let me start by confirming that these phones will not work with the majority of UK VoIP providers. After extensive testing, I couldn't get one to register with any of the UK VoIP providers I use. E.g. Voipfone, Voicehost, Sipgate, Voipcheap etc...
Cisco has a different way of handling SIP connections through NAT and a firewall on these phones, and it's not compatible with the way the rest of the world (including Asterisk) do it. This odd behavior is RFC compliant but highly non-standard and breaks symmetric NAT traversal workarounds commonly deployed by VOIP providers. Cisco support reports that this use of random high number ports to send SIP messages is a "security enhancement" compared with Cisco's other/older products.
All is not lost though, you can configure this phone for use with a local PBX such as FreePBX by disabling NAT on both the phone and the extension it's registered against.
1. Avoid the version 9 firmware, it's problematic, laggy and registrations sometimes fail. The most problematic for me was the v9.3.1 SR4 release, the most stable being v8.5.4. ***SO USE v8.5.4!!!***
2. There are three additional configuration files required to be placed in your TFTP folder along side the required firmware files. As above, I strongly recommend that you install and use SIP firmware v8.5.4. The following configuration files work perfectly with that release:
XMLDefault.cnf.xml
The contents of my TFTP folder based on the above look like this:
5. Now reboot you phone and make some calls!
Cisco has a different way of handling SIP connections through NAT and a firewall on these phones, and it's not compatible with the way the rest of the world (including Asterisk) do it. This odd behavior is RFC compliant but highly non-standard and breaks symmetric NAT traversal workarounds commonly deployed by VOIP providers. Cisco support reports that this use of random high number ports to send SIP messages is a "security enhancement" compared with Cisco's other/older products.
All is not lost though, you can configure this phone for use with a local PBX such as FreePBX by disabling NAT on both the phone and the extension it's registered against.
1. Avoid the version 9 firmware, it's problematic, laggy and registrations sometimes fail. The most problematic for me was the v9.3.1 SR4 release, the most stable being v8.5.4. ***SO USE v8.5.4!!!***
2. There are three additional configuration files required to be placed in your TFTP folder along side the required firmware files. As above, I strongly recommend that you install and use SIP firmware v8.5.4. The following configuration files work perfectly with that release:
XMLDefault.cnf.xml
Code: Select all
SEP[MAC].cnf.xml
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation6 model="IP Phone 7970">SIP70.8-5-4S</loadInformation6>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
Code: Select all
dialplan.xml
<device>
<fullConfig>true</fullConfig>
<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>46.31.225.134</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>[IPADDRESS]</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>0</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-5-4S</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>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:30</displayOnTime>
<displayOnDuration>00:10</displayOnDuration>
<displayIdleTimeout>00:59</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<networkLocale>United_Kingdom</networkLocale>
<networkLocaleInfo>
<name>United_Kingdom</name>
<version>10.5.3.0</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleTimeout>0</idleTimeout>
<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>[IPADDRESS]</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>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>60</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>g711a</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>32766</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>[USERID]</phoneLabel>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>[USERID]</featureLabel>
<name>[USERID]</name>
<displayName>[USERID]</displayName>
<contact>[USERID]</contact>
<proxy>[IPADDRESS]</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>[USERID]</authName>
<authPassword>[PASSWORD]</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>
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IMPORTANT: Each phone should have its own configuration file so you need to rename the file 'SEP[MAC].cnf.xml'. Simply replace “[MAC]” with the MAC address of your Cisco 7970. E.g. If your phones MAC address is '01:02:03:AB:CD:0A', rename the file 'SEP[MAC].cnf.xml' to 'SEP010203ABCD0A.cnf.xml'. FYI, the filename of the required v8.5.4 SIP firmware is 'cmterm-7970_7971-sip.8-5-4.zip'.<DIALTEMPLATE>
<TEMPLATE MATCH="999" Timeout="0"/> <!-- Emergency -->
<TEMPLATE MATCH="112" Timeout="0"/> <!-- Emergency -->
<TEMPLATE MATCH="101" Timeout="0"/> <!-- Almost an Emergency -->
<TEMPLATE MATCH="111" Timeout="0"/> <!-- NHS emergency and urgent care services -->
<TEMPLATE MATCH="100" Timeout="0"/> <!-- Operator -->
<TEMPLATE MATCH="155" Timeout="0"/> <!-- International Operator -->
<TEMPLATE MATCH="123" Timeout="0"/> <!-- Speaking Clock -->
<TEMPLATE MATCH="1471" TIMEOUT="0"/> <!-- Call Return -->
<TEMPLATE MATCH="1571" Timeout="0"/> <!-- Voicemail -->
<TEMPLATE MATCH="1572" Timeout="0"/> <!-- Group Voicemail -->
<TEMPLATE MATCH="118..." Timeout="0"/> <!-- Men with moustaches -->
<TEMPLATE MATCH="116..." Timeout="0"/> <!-- Pan-European Social Help -->
<TEMPLATE MATCH="08001111" Timeout="0"/> <!-- Childline -->
<TEMPLATE MATCH="0845464." Timeout="0"/> <!-- NHS Direct -->
<TEMPLATE MATCH="0500......" Timeout="0"/> <!-- Apparently 0500 is always 10 digits -->
<!-- Uncomment if you care about this one. Worst case, you'll wait 5 seconds -->
<!-- <TEMPLATE MATCH="016977...." Timeout="0"/> --> <!-- Brampton, Carlisle. Also 10 digits -->
<TEMPLATE MATCH="00*" Timeout="5"/> <!-- International, 00 prefixed. No fixed length -->
<TEMPLATE MATCH="0.........." Timeout="0"/> <!-- UK 11 digit, 0 prefixed -->
<TEMPLATE MATCH="1410.........." Timeout="0"/> <!-- UK 11 digit, 141 prefixed -->
<TEMPLATE MATCH="14700.........." Timeout="0"/> <!-- UK 11 digit, 1470 prefixed -->
<TEMPLATE MATCH="\*.." Timeout="0"/> <!-- Asterisk *.. codes -->
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>
The contents of my TFTP folder based on the above look like this:
- United_Kingdom/g3-tones.xml
- apps70.8-5-4TH1-6.sbn
- cnu70.8-5-4TH1-6.sbn
- cvm70sip.8-5-4TH1-6.sbn
- dialplan.xml
- dsp70.8-5-4TH1-6.sbn
- jar70sip.8-5-4TH1-6.sbn
- load119.txt
- load30006.txt
- SEP010203ABCD0A.cnf.xml ***This was the SEP[MAC].cnf.xml file, I renamed it with the MAC address of my phone!***
- SIP70.8-5-4S.loads
- term70.default.loads
- term71.default.loads
- XMLDefault.cnf.xml
- [IPADDRESS] - Replace all of this text with the IP address of your PBX. E.g. 192.168.1.10
- [USERID] - Replace all of this text with your extension username/userid. E.g. 200
- [PASSWORD] - Replace all of this text with your extension secret/password. E.g. password
- NAT Mode = No- (no)
- Qualify = no
5. Now reboot you phone and make some calls!
paulash134 liked this
- Appreciate my support? Feel free to buy me a coffee.
- Voipfone are offering you the chance to trial their VoIP service for free for 30 days. Sign me up!
- Tired of shared hosting? You're not alone! Grab a high performance server and get $100 in free credit. Sign me up!
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