[AsteriskBrasil] Autenticação Cisco 7942
Rafael Ribeiro - iPhone
rafaelribeiro.sp em gmail.com
Domingo Fevereiro 1 21:59:20 BRST 2015
Teste desabilitando o proxy.
Alguma mensagem aparece no cli?
Rafael Ribeiro
Sent from iPhone
> Em 01/02/2015, às 19:57, Luiz Eduardo F. Sampaio <rede.computador at gmail.com> escreveu:
>
> Voce ja viu permissao na pasta do tftp do servidor?
>
> Luiz Eduardo F. Sampaio
>
> Em 01/02/2015 18:53, Luis Carlos Fidalgo escreveu:
>> Boa tarde amigos,
>> Todos nossos telefones são Cisco 7942 e funcionam muito bem em um elastix 2.4.
>> Estamos fazendo uns testes na versão 3.0 e não conseguimos mais autenticar nossos telefones, mudamos a forma de autenticação que é diferente das versões anteriores do elastix, mas mesmo assim não autentica, me parece que nem chga a consultar nada no servidor.
>> Alguém tem alguma dica,
>>
>> Segue meu SEP….
>>
>> Servidor Elastix MT 3.0: 192.168.0.200
>> Organização: intra.nexlayer.net
>>
>>
>>
>> <device>
>> <deviceProtocol>SIP</deviceProtocol>
>> <sshUserId>cisco</sshUserId>
>> <sshPassword>cisco</sshPassword>
>> <devicePool>
>> <dateTimeSetting>
>> <dateTemplate>D/M/AA</dateTemplate>
>> <timeZone>South America Standard/Daylight Time</timeZone>
>> <ntps>
>> <ntp>
>> <name>192.168.0.200</name>
>> <ntpMode>unicast</ntpMode>
>> </ntp>
>> </ntps>
>> </dateTimeSetting>
>> <callManagerGroup>
>> <members>
>> <member priority="0">
>> <callManager>
>> <ports>
>> <ethernetPhonePort>2000</ethernetPhonePort>
>> <sipPort>5060</sipPort>
>> <securedSipPort>5061</securedSipPort>
>> </ports>
>> <processNodeName>192.168.0.200</processNodeName>
>> </callManager>
>> </member>
>> </members>
>> </callManagerGroup>
>> </devicePool>
>> <sipProfile>
>> <sipProxies>
>> <backupProxy></backupProxy>
>> <backup></backup>
>> <emergencyProxy></emergencyProxy>
>> <emergency></emergency>
>> <outboundProxy></outboundProxy>
>> <outbound></outbound>
>> <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>1</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>
>> <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>
>> <phoneLabel>NEXLAYER</phoneLabel>
>> <stutterMsgWaiting>0</stutterMsgWaiting>
>> <callStats>false</callStats>
>> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
>> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
>> <startMediaPort>16384</startMediaPort>
>> <stopMediaPort>32766</stopMediaPort>
>> <sipLines>
>> <line button="1">
>> <featureID>9</featureID>
>> <featureLabel>2000</featureLabel>
>> <proxy>192.168.0.200</proxy>
>> <port>5060</port>
>> <name>2000</name>
>> <displayName>2000-1</displayName>
>> <autoAnswer>
>> <autoAnswerEnabled>2</autoAnswerEnabled>
>> </autoAnswer>
>> <callWaiting>3</callWaiting>
>> <authName>2000 at intra.nexlayer.net</authName>
>> <authPassword>nossa senha</authPassword>
>> <sharedLine>false</sharedLine>
>> <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
>> <messagesNumber>*97</messagesNumber>
>> <ringSettingIdle>4</ringSettingIdle>
>> <ringSettingActive>5</ringSettingActive>
>> <contact>2000</contact>
>> <forwardCallInfoDisplay>
>> <callerName>true</callerName>
>> <callerNumber>true</callerNumber>
>> <redirectedNumber>false</redirectedNumber>
>> <dialedNumber>true</dialedNumber>
>> </forwardCallInfoDisplay>
>> </line>
>>
>>
>> <line button="2">
>> <featureID>9</featureID>
>> <featureLabel>2001</featureLabel>
>> <proxy>192.168.0.200</proxy>
>> <port>5060</port>
>> <name>2001</name>
>> <displayName>2001-2</displayName>
>> <autoAnswer>
>> <autoAnswerEnabled>2</autoAnswerEnabled>
>> </autoAnswer>
>> <callWaiting>3</callWaiting>
>> <authName>2001 at intra.nexlayer.net</authName>
>> <authPassword>nossa senha</authPassword>
>> <sharedLine>false</sharedLine>
>> <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
>> <messagesNumber>*97</messagesNumber>
>> <ringSettingIdle>4</ringSettingIdle>
>> <ringSettingActive>5</ringSettingActive>
>> <contact>2001</contact>
>> <forwardCallInfoDisplay>
>> <callerName>true</callerName>
>> <callerNumber>true</callerNumber>
>> <redirectedNumber>false</redirectedNumber>
>> <dialedNumber>true</dialedNumber>
>> </forwardCallInfoDisplay>
>> </line>
>>
>>
>> </sipLines>
>> <voipControlPort>5060</voipControlPort>
>> <dscpForAudio>184</dscpForAudio>
>> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
>> <dialTemplate>dialplan.xml</dialTemplate>
>> </sipProfile>
>> <commonProfile>
>> <phonePassword></phonePassword>
>> <backgroundImageAccess>true</backgroundImageAccess>
>> <callLogBlfEnabled>1</callLogBlfEnabled>
>> </commonProfile>
>> <loadInformation>SIP42.8-5-3S</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>
>> <spanToPCPort>1</spanToPCPort>
>> <loggingDisplay>1</loggingDisplay>
>> <loadServer></loadServer>
>> </vendorConfig>
>> <versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
>> <networkLocale>US</networkLocale>
>> <networkLocaleInfo>
>> <name>US</name>
>> <version>5.0(2)</version>
>> </networkLocaleInfo>
>> <deviceSecurityMode>1</deviceSecurityMode>
>> <authenticationURL></authenticationURL>
>> <directoryURL></directoryURL>
>> <idleURL></idleURL>
>> <informationURL></informationURL>
>> <messagesURL></messagesURL>
>> <proxyServerURL></proxyServerURL>
>> <servicesURL></servicesURL>
>> <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>
>> </device>
>>
>>
>>
>>
>>
>> Obrigado,
>>
>>
>>
>> Luis Carlos
>>
>>
>>
>> _______________________________________________
>> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
>> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
>> Intercomunicadores para acesso remoto via rede IP e telefones IP
>> Conheça todo o portfólio em www.Khomp.com
>> _______________________________________________
>> ALIGERA – Fabricante e desenvolvedor nacional de Soluções para telefonia IP .
>> Gateway Sip, Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
>> Banco de Canais Analógicos – Appliance Asterisk Acesse www.aligera.com.br
>> _______________________________________________
>> DIGIVOICE: Fabricante pioneiro em Banco de Canais e Placas E1, GSM, FXO e FXS para Asterisk e Elastix. Temos Cursos de Telefonia IP e Asterisk.
>> Construa soluções de PABX IP com produtos DigiVoice - visite www.digivoice.com.br
>> _______________________________________________
>> Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe at listas.asteriskbrasil.org
>
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
> Intercomunicadores para acesso remoto via rede IP e telefones IP
> Conheça todo o portfólio em www.Khomp.com
> _______________________________________________
> ALIGERA – Fabricante e desenvolvedor nacional de Soluções para telefonia IP .
> Gateway Sip, Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Banco de Canais Analógicos – Appliance Asterisk Acesse www.aligera.com.br
> _______________________________________________
> DIGIVOICE: Fabricante pioneiro em Banco de Canais e Placas E1, GSM, FXO e FXS para Asterisk e Elastix. Temos Cursos de Telefonia IP e Asterisk.
> Construa soluções de PABX IP com produtos DigiVoice - visite www.digivoice.com.br
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe at listas.asteriskbrasil.org
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