CCNA Blog: Cisco IP Phone Configuration with Asterisk

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Wednesday 29 February 2012

Cisco IP Phone Configuration with Asterisk

Getting the Cisco IP Phone 7970 G to work together with the software PBX Asterisk was something I had my hands on a couple of years back. Here’s how you can get them talking together.
You need a couple of things to get this working:

A functioning DHCP server
A functioning TFTP server
SIP Firmware from Cisco This is just a gzipped and tar’ed file.
A functioning asterisk server
A Cisco IP Phone

According to a recent installation, the TFTP server must contain the following files

apps70.1-1-2-26.sbn
cnu70.3-1-2-26.sbn
cvm70sip.8-0-2-25.sbn
dsp70.1-1-2-26.sbn
jar70sip.8-0-2-25.sbn
SIP70.8-0-3S.loads
term70.default.loads
term71.default.loads
SEP<MACADDRESS>.cnf.xml

The file you should pay the most attention to is the SEP<MACADDRESS>.cnf.XML file, this is the configuration file. The configuration file is in XML format. You can find a sample configuration here that should work.

<device xsi:type=”axl:XIPPhone” ctiid=”203849429″ uuid=”{96f8508b-10ef-f98c-d20d-0471777ec725}”>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId></sshUserId>
<sshPassword></sshPassword>
<devicePool uuid=”{a755aa55-089c-2b47-9603-c7d51b9ca4b5}”>
<dateTimeSetting uuid=”{9ec4850a-7748-11d3-bdf0-00108302ead1}”>
<dateTemplate>M/D/Y</dateTemplate>
<timeZone>Greenwich Standard Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority=”0″>
<callManager>
<name>ccm-beta-5-1</name>
<description>CallManager 5.0 Beta Pub – 5.0.1.032</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>ccm-beta-5-1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid=”{cd241e11-4a58-4d3d-9661-f06c912a18a3}”>
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1></ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1>IP ADDRESS TO SIP SERVER</sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
Default
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy>USECALLMANAGER</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-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>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
none
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button=”1″>
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3302</name>
<displayName>3302</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName></authName>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>7b452e87-4496-4762-e11f-b26751a1884b</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate></dialTemplate>
<softKeyFile>SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml</softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-0-3S</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><daysDisplayNotActive>1,7</daysDisplayNotActive><displayOnTime>07:30</displayOnTime><displayOnDuration>10:30</displayOnDuration><displayIdleTimeout>01:00</displayIdleTimeout><spanToPCPort>1</spanToPCPort></vendorConfig>
<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>
<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>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://ccm-beta-5-1:8080/ccmcip/authenticate.jsp</authenticationURL>
<directoryURL>http://ccm-beta-5-1:8080/ccmcip/xmldirectory.jsp</directoryURL>
<idleURL></idleURL>
<informationURL>http://ccm-beta-5-1:8080/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://10.86.5.102/CiscoServices/index.xml</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>ccm-beta-5-1</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<line button=”3″>
<featureID>2</featureID>
<featureLabel>2000</featureLabel>
<speedDialNumber>2000</speedDialNumber>
</line>
<natReceivedProcessing>true</natReceivedProcessing>
<natEnabled>true</natEnabled>
<natAddress></natAddress>
<dialTemplate>dialplan.xml</dialTemplate>
</device>

On the Asterisk server, you will have a file named sip.conf and to have the Cisco IP Phone talking to Asterisk you need this

[999999999]
username=999999999
type=friend
secret=password
nat=no
host=dynamic
canreinvite=no
dtmfmode=rfc2833
context=incoming
qualify=yes
disallow=all
allow=ulaw

That should be it, good luck!