Firmware of Cisco Phone

Question

i want to configure IP phone Cisco DX650 with Sip.Anyone have idea about it how i can upload firmware of sip feature

Answer 1

You need to supply the phone with a config file over TFTP, either by providing the right options with DHCP, or by setting the TFTP alternate setting via the menu. This settings makes the phone ask for a configuration file from the specified TFTP server, which you can easily host on your computer.

The config file needs to be named SEPXXXXXXXXXXXX.conf.xml where XXXXXXXXXX designates the MAC address of the phone.

I won't go into what settings should be in this file, but here's an example of a configuration for an asterisk system:

<?xml version="1.0" encoding="UTF-8"?>
<device  xsi:type="axl:XIPPhone" ctiid="62943" uuid="{e045c922-43ad-2320-24c9-be1f8abc3d0b}">
<fullConfig>true</fullConfig>
<portalDefaultServer></portalDefaultServer>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>YOUR USERNAME</sshUserId>
<sshPassword>YOUR PASSWORD</sshPassword>
<ipAddressMode>0</ipAddressMode>
<redirectEnable>false</redirectEnable>
<echoMultiEnable>false</echoMultiEnable>
<ipPreferenceModeControl>0</ipPreferenceModeControl>
<ipMediaAddressFamilyPreference>0</ipMediaAddressFamilyPreference>
<mlppDomainId>000000</mlppDomainId>
<mlppIndicationStatus>Off</mlppIndicationStatus>
<preemption></preemption>
<executiveOverridePreemptable></executiveOverridePreemptable>
<devicePool  uuid="{d0181915-1eac-910c-3a0f-f03c26afd832}">
<revertPriority>0</revertPriority>
<name>Phones - 1.5M Video EST EDT</name>
<dateTimeSetting  uuid="{daaf53f2-bb03-b274-953c-5090869fc211}">
<name>EST-5</name>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<olsonTimeZone>America/New_York</olsonTimeZone>
</dateTimeSetting>
<callManagerGroup>
<name>VOIP</name>
<tftpDefault>false</tftpDefault>
<members>
<member  priority="0">
<callManager>
<name>VOIP.ms</name>
<description></description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>INSERT YOUR ASTERISK OR VOIP ADDRESS</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></sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<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>
<URIDialingDisplayPreference>1</URIDialingDisplayPreference>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>60</timerRegisterExpires>
<timerRegisterDelta>0</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>
<kpml>3</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<T302Timer>5000</T302Timer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<poundEndOfDial>false</poundEndOfDial>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<organizationTopLevelDomain>YOUR IP ADDRESS OF ASTERISK OR VOIP PROVIDER</organizationTopLevelDomain>
<sipLines>
<line  button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>Office</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>201</name>
<displayName>YOUR DISPLAY NAME</displayName>
<autoAnswer>
<autoAnswerEnabled>0</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>YOUR USERNAME</authName>
<authPassword>YOUR PASSWORD</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>0</messageWaitingAMWI>
<messagesNumber>YOUR VOICEMAIL NUMBER</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact></contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>10</maxNumCalls>
<busyTrigger>6</busyTrigger>
</line>
<line  button="3">
<featureID>1</featureID>
</line>
</sipLines>
<externalNumberMask>YOUR DID OR PHONE NUMBER</externalNumberMask>
<voipControlPort>5060</voipControlPort>
<ringSettingBusyStationPolicy>1</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>SKd7581e75-e2ff-277e-6fcc-2a7739543647.xml</softKeyFile>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
</sipProfile>
<MissedCallLoggingOption>10</MissedCallLoggingOption>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
<loadInformation>sipdx650.10-2-5-194</loadInformation>
<inactiveLoadInformation></inactiveLoadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<allowBTContactImport>1</allowBTContactImport>
<allowBTMobileHandsfree>1</allowBTMobileHandsfree>
<recordingTone>0</recordingTone>
<settingsAccess>1</settingsAccess>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<deviceUIProfile>0</deviceUIProfile>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<garp>1</garp>
<multiUser>0</multiUser>
</vendorConfig>
<commonConfig>
<ciscoCamera>1</ciscoCamera>
<videoCapability>0</videoCapability>
<webProtocol>0</webProtocol>
<webAccess>0</webAccess>
<sshAccess>0</sshAccess>
<sendKeyAction>1</sendKeyAction>
<RingLocale>0</RingLocale>
<appInstallFromAndroidMarket>true</appInstallFromAndroidMarket>
</commonConfig>

<versionStamp>1387322115-49d5fd49-52b6-4926-b708-11c02cb22c22</versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version></version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>Canada</networkLocale>
<networkLocaleInfo>
<name>Canada</name>
<uid>64</uid>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>

<transportLayerProtocol>1</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<phonePersonalization>1</phonePersonalization>
<rollover>0</rollover>
<singleButtonBarge>0</singleButtonBarge>
<joinAcrossLines>0</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>YOUR ASTERISK OR VOIP ADDRESS</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<advertiseG722Codec>0</advertiseG722Codec>
<mobility>
<handoffdn>8888</handoffdn>
<dtmfdn>41200</dtmfdn>
<ivrdn>86547810</ivrdn>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<TLSResumptionTimer>0</TLSResumptionTimer>
<phoneServices  useHTTPS="true">
<provisioning>0</provisioning>
<phoneService  type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService  type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService  type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService  type="1" category="0">
<name>Personal Directory</name>
<url>Application:Cisco/PersonalDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>

For more information regarding the DX650, that does seem to support SIP without any special firmware, look here: https://community.cisco.com/t5/ip-telephony-and-phones/configure-cisco-dx650-with-asterisk-sip-provider/td-p/3991587