Olá, As permissões estão ok, pois esta carregando normalmente as novas configurações no telefone, a questão é que não autentica a conta mesmo
Estou fazendo outras mudanças para ver. Luis De: asteriskbrasil-boun...@listas.asteriskbrasil.org [mailto:asteriskbrasil-boun...@listas.asteriskbrasil.org] Em nome de Luiz Eduardo F. Sampaio Enviada em: domingo, 1 de fevereiro de 2015 19:57 Para: asteriskbrasil@listas.asteriskbrasil.org Assunto: Re: [AsteriskBrasil] Autenticação Cisco 7942 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</silentPeriodBetweenCallWaitingBurs ts> <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>2...@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>2...@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-unsubscr...@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-unsubscr...@listas.asteriskbrasil.org