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Re: Cisco 7941G и Asterisk 13.13
Добавлено: 25 янв 2017, 12:41
pawuwa
Я пытаюсь набрать номер 210 вообще любой номер, и после первой цифры в трубке занято. Все буду пробовать по Вашему совету, провайдера нет все в одной подсети.
Re: Cisco 7941G и Asterisk 13.13
Добавлено: 25 янв 2017, 14:29
ded
Вам надо вылепить чуткими пальцами файл dialplan.xml, клоторый засосёт телефон из /tftpboot при загрузке, если у вас всё правильно организовано. Примеры - в сети.
Если неправильно организовано - ещё на три страницы пустой переписки в форуме будет.
Re: Cisco 7941G и Asterisk 13.13
Добавлено: 25 янв 2017, 14:49
pawuwa
Я использую вот это и думаю что все должно быть ок. Судя по Вашим словам ded это не вариант. Сейчас залью dialplan.xml.
<dialTemplate>
<TEMPLATE MATCH="*" Timeout="3"/> <!-- Anything else -->
</dialTemplate>
Re: Cisco 7941G и Asterisk 13.13
Добавлено: 25 янв 2017, 15:56
pawuwa
Все заработало. Всем ОГРОМНОЕ спасибо.
Моя полная схема Hyper-V Server 2012R2 > Ubuntu 14.02 > Asterisk 13.13.0 (FreePBX).
Телефон - Cisco 7941G
Прошивка - SIP41.8-5-4S
Конфиг телефона SEPMAC.cnf
<?xml version="1.0" ?>
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshAccess>0</sshAccess>
<sshUserId></sshUserId>
<sshPassword></sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.Y</dateTemplate>
<timeZone>Russian Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>ASTERISK-IP</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>ASTERISK-IP</name>
<description>CallManager 5.0</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>SIP-PORT</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>ASTERISK-IP</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>0</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP41.8-5-4S</loadInformation>
<loadInformation434 model="Cisco 7941">SIP41.8-5-4S</loadInformation434>
<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>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>10:30</displayOnTime>
<displayOnDuration>06:05</displayOnDuration>
<displayIdleTimeout>00:05</displayIdleTimeout>
<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<userLocale>
<name>Russian_Russian_Federation</name>
<uid></uid>
<langCode>ru_RU</langCode>
<version>8.4.3.1000-1</version>
<winCharSet>utf-8</winCharSet>
</userLocale>
<networkLocale>Russian_Federation</networkLocale>
<networkLocaleInfo>
<name>Russian_Federation</name>
<uid></uid>
<version>8.4.3.1000-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<directoryURL></directoryURL>
<servicesURL></servicesURL>
<idleURL></idleURL>
<messagesURL></messagesURL>
<proxyServerURL>ASTERISK-IP</proxyServerURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy>ASTERISK-IP</backupProxy>
<backupProxyPort>SIP-PORT</backupProxyPort>
<emergencyProxy>ASTERISK-IP</emergencyProxy>
<emergencyProxyPort>SIP-PORT</emergencyProxyPort>
<outboundProxy>ASTERISK-IP</outboundProxy>
<outboundProxyPort>SIP-PORT</outboundProxyPort>
<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>
<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>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>true</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32768</stopMediaPort>
<voipControlPort>SIP-PORT</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>Cisco</phoneLabel>
<natReceivedProcessing>false</natReceivedProcessing>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>ASTERISK-EXTENSION</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>SIP-PORT</port>
<name>ASTERISK-EXTENSION</name>
<displayName>ASTERISK-EXTENSION</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>ASTERISK-EXTENSION</authName>
<authPassword>EXTENSION-PASSWORD</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>ASTERISK-EXTENSION</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
</device>
dialplan.xml
<DIALTEMPLATE>
<TEMPLATE MATCH="." TIMEOUT="15" User="Phone"/>
<TEMPLATE MATCH="...." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="2.." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="9......." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="13...." TIMEOUT="2" User="Phone"/>
<TEMPLATE MATCH="02........" TIMEOUT="2" User="Phone"/>
</DIALTEMPLATE>