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Hlavní témata => Sítě => Téma založeno: Tomas Srna 04. 04. 2011, 20:39:34
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Ahoj
Mam problem s Cisco 7911g telefonom, cez tftp sa nahra konfigurak, v menu vidim nahranu konfiguraciu, ale nechce sa pripojit na SIP server Asterisk. Na asterisku bezi TCP aj UDP.
Napriek spravnej konfiguracii ale telefon pise "Registering..." a z tohto stavu sa nikdy nedostane.
Neviete kde by mohol byt problem?
Dakujem
Tomas Srna
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sniffni si komunikaci mezi IP telefonem a asteriskem..
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Mozno blba otazka, ale mas tam SIP firmware? Standartne je tam SCCP. Ak hej tak si nastav span na PC port a urob data sniff(alebo na asterisk servri ak mozes).
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prave "Registering.." by melo znacit ze tam neni MGCP.. ale SIP
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MGCP sa pouziva medzi CUCM(PBX) a voice gateway nie medzi telefonom a PBX (v cisco svete) SCCP/SIP je len rozdielny signaling bud na porte 2000 pre SCCP alebo 5060 sip. Aj 7911 so SCCP firmwarom bude ukazovat registering ked sa z nejakeho dovodu(DNS) nevie pripojit na PBX.
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Před několika lety jsem s tím také bojoval, telefon si stáhl konfiguraci, ale nechtěl se registrovat. Bylo to tou konfigurací... Stačila malá změna zdánlivě nevinného parametru a... Telefon se buď registroval a nebo naopak odmítal registrovat. Několikadenní laborací jsme našli funkční konfiguraci na které telefony jedou dodnes.
Spousta informací je na http://www.voip-info.org/ (http://www.voip-info.org/) - hledej "Asterisk 7911"
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Tak som si to skusil sam. Ked som pouzil hociaky firmware z 9. rady tak sa telefon any nesnazil zaregistrovat. Ked som pouzil 8.5.4 tak to islo v pohode.
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Tu je config pre 7945
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.Y</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>{TBexternalIPaddress}</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>172.16.101.62</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>0</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP45.8-5-4S</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>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>10:30</displayOnTime>
<displayOnDuration>06:05</displayOnDuration>
<displayIdleTimeout>00:05</displayIdleTimeout>
<webAccess>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<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>1</deviceSecurityMode>
<authenticationURL>http://{TBexternalIPaddress}/xmlservices/authentication.php</authenticationURL>
<directoryURL>http://{TBexternalIPaddress}/xmlservices/PhoneDirectory.php</directoryURL>
<idleTimeout>0</idleTimeout>
<idleURL></idleURL>
<informationURL>http://{TBexternalIPaddress}/xmlservices/index.php</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://{TBexternalIPaddress}/xmlservices/index.php</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>
<sipProfile>
<sipProxies>
<backupProxy>172.16.101.62</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></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>true</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>0</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>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711u</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>Asterisk 6000</phoneLabel>
<natEnabled>true</natEnabled>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>6000</featureLabel>
<name>6000</name>
<displayName>6000</displayName>
<contact>6000</contact>
<proxy>172.16.101.62</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>6000</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>8888</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>9</featureID>
<featureLabel>6001</featureLabel>
<name>6001</name>
<displayName>6001</displayName>
<contact>6001</contact>
<proxy>172.16.101.62</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>6001</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>8888</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
</device>
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Skusal som ten configurak, ale pise mi UNPROVISIONED po jeho nahrati.
Dalej: Mate niekto konkretnu skusenost presne s 7911tkou? Mne kamarat znaly tychto telefonov povedal, ze maju nejaky "osekany" SIP protokol, vraj aby motivovali ludi si tam dat CallManager. Neviete o tom nieco?
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byla chyba pořizovat cisco telefon, protože ten je určen pro provoz s callmanagenerem!