[AsteriskBrasil] RES: Autenticação Cisco 7942

Luis Carlos Fidalgo luisfidafer em gmail.com
Segunda Fevereiro 2 00:46:32 BRST 2015


Boa noite,

Vou fazer isso amanha e testar.

 

 

Obrigado,

 

 

 

Luis

 

 

De: asteriskbrasil-bounces at listas.asteriskbrasil.org [mailto:asteriskbrasil-bounces at listas.asteriskbrasil.org] Em nome de Rafael Ribeiro - iPhone
Enviada em: domingo, 1 de fevereiro de 2015 21:59
Para: asteriskbrasil at listas.asteriskbrasil.org
Assunto: Re: [AsteriskBrasil] Autenticação Cisco 7942

 

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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