Проблема выражается следующим образом:
1)На экране есть надпись "unprovisioned"
2) Телефон подключается к астериску, на него приходят звонки, но при поднятии трубки ничего не происходит, телефон продолжает звонить.
-Лечиться правильным конфигом
*Версия прошивки:SIP45,8-5-4S
*Актуально для моделей: Cisco 79XX( Unified IP Phone 7900 Series )
*Пример файла SEP*mac*.cnf.xml (например SEP001E13AFB010.cnf.xml)
по ссылке http://pastecode.ru/39d2d1/
Конечно его надо подправить под ваш USER_ID и USER_PASSWORD
На всякий случай запилю содержания файла здесь:
1)На экране есть надпись "unprovisioned"
2) Телефон подключается к астериску, на него приходят звонки, но при поднятии трубки ничего не происходит, телефон продолжает звонить.
-Лечиться правильным конфигом
*Версия прошивки:SIP45,8-5-4S
*Актуально для моделей: Cisco 79XX( Unified IP Phone 7900 Series )
*Пример файла SEP*mac*.cnf.xml (например SEP001E13AFB010.cnf.xml)
по ссылке http://pastecode.ru/39d2d1/
Конечно его надо подправить под ваш USER_ID и USER_PASSWORD
На всякий случай запилю содержания файла здесь:
<!-- FIXME: Change to your own phone number (or another unique ID) -->
<device xsi:type="axl:XIPPhone" ctiid="USER_ID">
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>default</sshUserId>
<sshPassword>user</sshPassword>
<devicePool>
<dateTimeSetting>
<!-- FIXME: Set your preferred date format and timezone here -->
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Ekaterinburg Standard Time</timeZone>
<ntps>
<!-- NTP might not actually work, but the phone can set the
date/time from the SIP response headers -->
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<!-- This section probably does not do anything useful. -->
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>127.0.0.1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<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>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>
<!-- Force short registration timeout to keep NAT connection alive -->
<timerRegisterExpires>180</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>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<!-- FIXME: This will appear in the upper right corner of the display -->
<phoneLabel>USER_ID</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>16391</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<!-- FIXME: Text to display next to line button #1 -->
<featureLabel>USER_ID</featureLabel>
<!-- FIXME: FQDN or IP of your SIP proxy -->
<proxy>192.168.1.100</proxy>
<port>5060</port>
<!-- FIXME: SIP username -->
<name>USER_ID</name>
<!-- FIXME: Name to display on outbound caller ID -->
<displayName>USER_ID</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<!-- FIXME: SIP authorization name (often matches username) -->
<authName>USER_ID</authName>
<!-- FIXME: SIP authorization password -->
<authPassword>USER_PASSWORD</authPassword>
<sharedLine>true</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<!-- FIXME: "Messages" key will autodial this number -->
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact></contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>2</featureID>
<featureLabel>ATT</featureLabel>
<speedDialNumber>18002255288</speedDialNumber>
</line>
<!-- FIXME: Add more line buttons or speed dial entries here -->
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<!-- FIXME: Change this to upgrade the firmware -->
<loadInformation>SIP45.8-5-4S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>1</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<!-- For Sunday (1) and Saturday (7):
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
Current default is to enable the display 24/7.
-->
<daysDisplayNotActive></daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>24:00</displayOnDuration>
<displayIdleTimeout>00:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp></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>0</deviceSecurityMode>
<!--
<authenticationURL>http://yourwebserver/authenticate.php</authenticationURL>
<directoryURL>http://yourwebserver/directory.xml</directoryURL>
-->
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<!--
<servicesURL>http://phone-xml.berbee.com/menu.xml</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>
</device>
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