CCNA Blog: Cisco IP Phone Configuration with Asterisk

Showing posts with label Cisco IP Phone Configuration with Asterisk. Show all posts

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!