我想用 Sip 配置 IP 电话 Cisco DX650。任何人都知道如何上传 sip 功能的固件
思科电话固件
网络工程
思科
思科-ios
网络电话
cisco-7900-ip 电话
2022-02-08 12:14:03
1个回答
您需要通过 TFTP 为电话提供配置文件,方法是提供正确的 DHCP 选项,或者通过菜单设置 TFTP 备用设置。此设置使电话从指定的 TFTP 服务器请求配置文件,您可以轻松地将其托管在您的计算机上。
配置文件需要命名为SEPXXXXXXXXXXXX.conf.xmlXXXXXXXXXX 指定电话的 MAC 地址。
我不会讨论这个文件中应该有哪些设置,但这里有一个星号系统的配置示例:
<?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>
有关 DX650 的更多信息,它似乎在没有任何特殊固件的情况下支持 SIP,请查看此处:https ://community.cisco.com/t5/ip-telephony-and-phones/configure-cisco-dx650-with-asterisk -sip-provider/td-p/3991587
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