Instrukcja demontażu i serwisowania Cisco Cisco 7960

Instrukcja demontażu i serwisowania Cisco Cisco 7960 jest dokumentem opracowanym dla serwisantów oferujących usługi w zakresie demontażu i serwisowania telefonów Cisco 7960. Wyjaśnia ona, jak prawidłowo rozmontować i regulować telefon, jak również, jakie narzędzia i części są potrzebne do wykonania tych czynności. Instrukcja wyjaśnia również, jakie kroki należy wykonać podczas wymiany komponentów i jakie mogą być konsekwencje niewłaściwego montażu. Instrukcja dostarcza informacji na temat tego, jakie procedury serwisowe są zalecane, aby zapewnić jak najlepszą jakość usług i zminimalizować ryzyko uszkodzenia sprzętu.

Ostatnia aktualizacja: Instrukcja demontażu i serwisowania Cisco Cisco 7960

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Quick Start Guide
Cisco 7911
IP Telephone

CONTENTS

CISCO 7911 IP PHONE FEATURE AND KEY OVERVIEW........................................... 3
BASIC FEATURES......................................................................................................... 5
Installing the IP Phone and Connecting to the Network......................................................................... 5
Placing a Call...................................................................................

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Setting up Voice Mail................................................................................................................................ 11 Accessing Voicemail................................................................................................................................ 11 Checking Voicemail from an Outside Line............................................................................................. 11 DIRECTORY FEATURES.........................................

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Cisco 7911 IP Phone Feature and Key Overview Handset with Indicator Light Functions like a traditional handset. The light strip at the top of the handset blinks when the phone rings and will be set to remain lit when you receive a voice mail message. LCD Screen Displays features such as time, date, your phone number, caller ID if available, line/call status and soft key tabs. Cisco IP Phone Model Type Indicates your Cisco IP phone model. “Soft Key” Buttons Four interactive “so
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Volume Button Increases or decreases the volume for the handset, headset or speakerphone (depending on which is being used). • Press the button to decrease the volume of the handset, speakerphone, or headset • Press the button to increase volume of the handset, speakerphone, or headset • To save the volume setting, press the Save soft key Navigation Button (Toggle UP/DOWN Key) or “SCROLL Key” Enables you to scroll through text and select features displayed on the LCD Screen Dial
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Basic Features Installing the IP Phone and Connecting to the Network 1. Connect an Ethernet cable to the Network port (left port) on the phone. The other end of the Ethernet cable must be plugged into a powered Ethernet jack. All VoIP-capable Ethernet jacks are marked with a red dot. 2. Connect the handset to its respective port. 3. Connect the Ethernet cable from your Desktop PC to the Access Port on the IP Phone (right port - optional). 4. Connect the power plug to the DC48V AC Ada
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Additional Call Handling Each phone line you have available can handle up to six individual calls. Therefore, it is possible that you will have a call in progress and receive another call on the same line. Or you could have a call in progress and decide to call someone else to confirm a piece of information, only to return to your original call. You can also use the additional call handling feature in order to set up a three- or four-way conference call. Answering an additional call w
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Soft Features Parking a Call Park a call when you want to store the call and then retrieve the call from another IP Phone. Call Park numbers are pre-configured by your system administrator. Follow the below steps to park a call: Step 1) During an active call, press the more soft key until you see the Park tab. Step 2) Press the Park tab soft key. The LCD screen displays a special call park number at which the call is stored. Step 3) Make a note of the call park number and then ha
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Forward All Calls Call Forwarding allows you to redirect all incoming calls from your Cisco IP Phone to another number. Step 1) Press the CFwdALL soft key (you will hear two beeps). Step 2) Dial the 4-digit extension to which all calls will be forwarded, OR Step 3) Dial the external number to which all calls will be forwarded (e. g., your cell phone). Dial the number exactly as you would normally dial it if you were placing a call. Note: After you enter the number, an animated phone ic
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Items to Consider for Conference Calls Once the conference call initiator disconnects, no new conferees can be added When you place a call on hold, the other callers will hear KUMR radio playing in the background. Consider using the mute instead of hold to avoid disrupting other callers. To abandon the addition of a conferee, press the Line button. IP Phones have a built-in conference call manager that can handle up to three parties simultaneously. If you need four parties on a line,
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Group Pickup (Optional) You can answer a call to your extension outside your call group. Step 1) As a phone rings at an extension outside of your call group, press a Line button. Step 2) Press the more soft key one time. Step 3) Press the GPickup soft key. Step 4) Dial the desired call pickup group number. Step 5) Answer the incoming call that is redirected to your phone. Redial the Last Number Called Redialing the most recently dialed number. Step 1) Lift the handset and press t
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Voicemail (optional) UMR subscribes to Unified Messaging from Cisco. Customers who currently have voicemail will automatically be converted over to the new Unified Messaging system at no extra charge. However, no voicemail messages will be transferred from the old system to the new system. Customers who do not have voicemail, but would like to subscribe to Unified Messaging should contact UMR Information Technology to establish a Unified Messaging account. That customer’s department
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Directory Features Call History Features Step 1) To view call histories, press the GLOBAL SERVICES button (right button next to the up/down SCROLL button). Step 2) Use the SCROLL button to select the Directories option and then press the Select soft key. Step 3) Use the SCROLL button to select the desired call history option: Missed Calls, Received Calls or Placed Calls. Step 4) Press the Select soft key to display the desired call history. Step 5) When you are finished, press the Exi
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Using the UMR Phone Directories Customers can use the built-in UMR phone directory to find any phone number on campus—VoIP and non-VoIP—as well as any individual’s number within the UM System. To find a departmental phone number on campus: Step 1) Press the GLOBAL SERVICES button. Step 2) Use the SCROLL button to select the Services option and press the Select soft key. Step 3) Use the SCROLL button to select the UMR Department Directory option on the menu. Step 4) Press the Select s
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Help and Settings Features LCD Display Contrast Step 1) Press the GLOBAL SERVICES button. Step 2) Use the SCROLL key to select the Settings option and press the Select soft key. Step 3) Use the SCROLL button to select the Contrast option on the Settings Menu. Step 4) Press the Select soft key to display the contrast settings. Step 5) Use the Up and Down soft keys to set the desired intensity of the LCD display and press the OK soft key to save your settings. Step 6) Press the Save soft
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Appendix: Fast Dials and Speed Dials In addition to all the standard features found within IP telephones, Cisco IP telephones allow customers to have two separate lists of Fast Dials and Speed Dials—separate but related entities. Fast Dials is a feature offered through the SERVICES button on all models of IP telephones. This is composed of a list of contacts whose numbers you would like to store on your phone for easy reference. By contrast, Speed Dials can be accessed simply by pre
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Configure your Cisco Personal Address Book From the Cisco Call Manager User Options Menu: 1. Click on Configure your Cisco Personal Address Book. Click Add a New Entry. Fill in the data fields with the requested information. a. First Name and Last Name are both required. b. If the contact is an on-campus employee, you should use their four-digit extension for their work number (e. g. 4357). c. Off-campus and long distance numbers should start with a “3”, especially if you
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Accessing Fast Dials through IP Phone Once you have subscribed and configured your Fast Dial numbers, you can access it on your IP phone in one of two ways, depending on the model of IP phone you are using: Model 7911: 1. Press the SERVICES button. Use the SCROLL button to select the “Services” option and press the Select soft key. Press the Select soft key to choose the “Fast Dials” option. Use the SCROLL button to select which Fast Dial number you need. 5. Press the
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S SP PE EE ED D D DIIA AL LS S Login to the Cisco CallManager at http://ccmuser. umr. edu. 1. Use your UMR userid. The default password is “cisco”. As with all passwords, we recommend that you change this password as soon as possible to one that matches UMR's strong password guidelines. Click on Add/Update your Speed Dials. For each Speed Dial index number, enter in the phone number you want. Note that the Display Text for most of the Speed Dial index numbers will not b
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Cisco SIP IP Phone 7960
Administrator Guide
Version 2. 0
Corporate Headquarters
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
USA
http://www. cisco. com
Tel: 408 526-4000
800 553-NETS (6387)
Fax: 408 526-4100
Customer Order Number: DOC-7810497=
Text Part Number: 78-10497-02

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THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE I
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CONTE N TS About This Guide ix Overview ix Who Should Use This Guide ix Objectives x Organization x Related Documentation xi Document Conventions xi Obtaining Documentation xv World Wide Web xv Documentation CD-ROM xv Ordering Documentation xv Obtaining Technical Assistance xv Cisco Connection Online xvi Technical Assistance Center xvi Documentation Feedback xvii 1 1-1 CHAPTER Product Overview What is Session Initiation Protocol? 1-1 Components of SIP 1-3 SIP Clients 1-4 SIP Servers 1-5 Cisco SI
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Contents What is the Cisco SIP IP Phone 7960? 1-5 Supported Features 1-7 Supported Protocols 1-10 Prerequisites 1-12 Cisco SIP IP Phone Connections 1-13 Connecting to the Network 1-13 Connecting to Power 1-14 Using a Headset 1-15 The Cisco SIP IP Phone with a Catalyst Switch 1-16 2 2-1 CHAPTER Getting Started with Your Cisco SIP IP Phone Initialization Process Overview 2-1 Installing the Cisco SIP IP Phone 2-3 Installation Task Summary 2-3 Downloading Files to Your TFTP Server 2-4 Configuring SI
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Contents Using the Cisco SIP IP Phone Menu Interface 2-21 Reading the Cisco SIP IP Phone Icons 2-22 Customizing the Cisco SIP IP Phone Ring Types 2-24 Creating Dial Plans 2-24 3 3-1 CHAPTER Managing Cisco SIP IP Phones Entering Configuration Mode 3-1 Unlocking Configuration Mode 3-2 Locking Configuration Mode 3-2 Modifying the Phone’s Network Settings 3-2 Modifying the Phone’s SIP Settings 3-5 Modifying SIP Parameters via a TFTP Server 3-8 Modifying the Default SIP Configuration File 3-8 Modifyi
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Contents APPENDIX A SIP Compliance with RFC-2543 Information A-1 SIP Functions A-2 SIP Methods A-2 SIP Responses A-3 1xx Response—Information Responses A-4 2xx Response—Successful Responses A-4 3xx Response—Redirection Responses A-5 4xx Response—Request Failure Responses A-5 5xx Response—Server Failure Responses A-10 6xx Response—Global Responses A-10 SIP Header Fields A-10 SIP Session Description Protocol (SDP) Usage A-12 APPENDIX B SIP Call Flows B-1 Call Flow Scenarios for Successful Calls B-
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Contents Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Busy) B-44 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No Answer) B-48 Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling B-52 Call Flow Scenarios for Failed Calls B-58 Gateway-to-Cisco SIP IP Phone—Called User is Busy B-58 Gateway-to-Cisco SIP IP Phone—Called User Does Not Answer B-60 Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error B-63 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Ca
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Contents GLOSSARY INDEX Cisco SIP IP Phone 7960 Administrator Guide viii 78-10497-02
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About This Guide Overview The Cisco Session Initiation Protocol (SIP) IP Phone 7960 Administrator Guide provides information about how to setup, connect cables to, and configure a Cisco SIP IP phone 7960 (hereafter referred to as a Cisco SIP IP phone). The administrator guide also provides information on how to configure the network and SIP settings and change the settings and options of the Cisco SIP IP phone. The administrator guide also includes reference information such as Cisco SIP IP
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About This Guide Objectives Objectives The Cisco SIP IP Phone 7960 Administrator Guide provides necessary information to get the Cisco SIP IP phone operational in a Voice-over-IP (VoIP) network. It is not the intent of this administrator guide to provide information on how to implement a SIP VoIP network. For information on implementing a SIP VoIP network, refer to the documents listed in the “Related Documentation” section on page xi. Organization This administrator guide is divided into
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About This Guide Related Documentation Related Documentation The following is a list of related Cisco SIP VoIP publications. For more information about implementing a SIP VoIP network refer to the following publications: • Session Initiation Protocol Gateway Call Flows • Session Initiation for VoIP on Cisco Access Platforms • Getting Started with the Cisco IP Phone 7960 • Installing the Wall Mount Kit for the Cisco IP Phone The following is a list of Cisco VoIP publications that provide inform
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About This Guide Document Conventions Notes use the following conventions: Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the publication. Cautions use the following conventions: Caution Means reader be careful. In this situation, you might do something that could result in equipment damage or loss of data. Warnings use the following conventions: Warning This warning symbol means danger. You are in a situation that could cause bodily in
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About This Guide Document Conventions Varoitus Tämä varoitusmerkki merkitsee vaaraa. Olet tilanteessa, joka voi johtaa ruumiinvammaan. Ennen kuin työskentelet minkään laitteiston parissa, ota selvää sähkökytkentöihin liittyvistä vaaroista ja tavanomaisista onnettomuuksien ehkäisykeinoista. (Tässä julkaisussa esiintyvien varoitusten käännökset löydät liitteestä "Translated Safety Warnings" (käännetyt turvallisuutta koskevat varoitukset). ) Attention Ce symbole d’avertissement indique un dang
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About This Guide Document Conventions Advarsel Dette varselsymbolet betyr fare. Du befinner deg i en situasjon som kan føre til personskade. Før du utfører arbeid på utstyr, må du være oppmerksom på de faremomentene som elektriske kretser innebærer, samt gjøre deg kjent med vanlig praksis når det gjelder å unngå ulykker. (Hvis du vil se oversettelser av de advarslene som finnes i denne publikasjonen, kan du se i vedlegget "Translated Safety Warnings" [Oversatte sikkerhetsadvarsler]. ) Avis
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About This Guide Obtaining Documentation Obtaining Documentation World Wide Web You can access the most current Cisco documentation on the World Wide Web at http://www. com, http://www-china. com, or http://www-europe. com. Documentation CD-ROM Cisco documentation and additional literature are available in a CD-ROM package, which ships with your product. The Documentation CD-ROM is updated monthly. Therefore, it is probably more current than printed documentation. The CD-ROM
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About This Guide Obtaining Technical Assistance Cisco Connection Online Cisco continues to revolutionize how business is done on the Internet. Cisco Connection Online is the foundation of a suite of interactive, networked services that provides immediate, open access to Cisco information and resources at anytime, from anywhere in the world. This highly integrated Internet application is a powerful, easy-to-use tool for doing business with Cisco. CCO’s broad range of features and services hel
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About This Guide Obtaining Technical Assistance To display the TAC web site that includes links to technical support information and software upgrades and for requesting TAC support, use www. com/techsupport. To contact by e-mail, use one of the following: Language E-mail Address English tac@cisco. com Hanzi (Chinese) chinese-tac@cisco. com Kanji (Japanese) japan-tac@cisco. com Hangul (Korean) korea-tac@cisco. com Spanish tac@cisco. com Thai thai-tac@cisco. com In North America, TAC can be reac
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About This Guide Obtaining Technical Assistance Cisco SIP IP Phone 7960 Administrator Guide xviii 78-10497-02
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CH APTER 1 Product Overview This chapter contains the following information about the Cisco SIP IP phone: • What is Session Initiation Protocol?, page 1-1 • What is the Cisco SIP IP Phone 7960?, page 1-5 • Prerequisites, page 1-12 • Cisco SIP IP Phone Connections, page 1-13 • The Cisco SIP IP Phone with a Catalyst Switch, page 1-16 What is Session Initiation Protocol? Session Initiation Protocol (SIP) is the Internet Engineering Task Force’s (IETF’s) standard for multimedia conferencing over IP
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Chapter 1 Product Overview What is Session Initiation Protocol? SIP provides the capabilities to: • Determine the location of the target end point—SIP supports address resolution, name mapping, and call redirection. • Determine the media capabilities of the target end point—Via Session Description Protocol (SDP), SIP determines the “lowest level” of common services between the end points. Conferences are established using only the media capabilities that can be supported by all end points. •

Aby skonfigurować Cisco 6921 pod SIP, musisz wykonać kilka kroków. Po pierwsze, pobierz najnowszą wersję oprogramowania wbudowanego dla telefonu i utwórz plik konfiguracyjny w oparciu o poniższą instrukcję. Gdy telefon pobierze informacje z serwera TFTP i będzie włączony, będzie gotowy do działania. Opiszmy wszystko krok po kroku.

1. Zainstaluj oprogramowanie tftp-server (na przykład Tftpd32). W ustawieniach oprogramowania wybierz folder z plikami konfiguracji i plikami oprogramowania wbudowanego. Wybierz interfejs sieci, do którego telefon jest podłączony.

2. Dokonaj konfiguracji oprogramowania wbudowanego tftp. Wejdź w menu, a następnie w ustawieniach kliknij w

Admin Settings > Network Setup > ipv4 setup > kliknij do dołu > Alternate tftp server > przełącz na YES > kliknij do dołu > tftp Server >

Ustawienia telefonu dla oprogramowania wbudowanego tftp. Aby dokonać konfiguracji, kliknij w: Admin Settings > Network Setup > ipv4 setup > kliknij do dołu > Alternate tftp server > zmień na YES >kliknij do doły > tftp Server > wprowadź adres IP Twojego komputera, na którym zainstalowany jest serwer tftp.

3. Struktura nazwy pliku konfiguracji telefonu wygląda następująco: SEP, następnie MAC-adres telefonu, a potem [. ]cnf[. ]xml (przykładowo: SEPD0D0FD17395E. cnf. xml).

Przykładowa zawartość pliku konfiguracyjnego:

<device><deviceProtocol>SIP</deviceProtocol><sshUserId>admin</sshUserId><sshPassword>Password</sshPassword><devicePool><dateTimeSetting><dateTemplate>D. M. Y</dateTemplate><timeZone>E. Europe Standard/Daylight Time</timeZone><ntps><ntp><name>time. windows. com</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>sip. zadarma. com</processNodeName></callManager></member></members></callManagerGroup></devicePool><sipProfile><sipProxies><backupProxy></backupProxy><backupProxyPort></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>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>true</remotePartyID><userInfo>None</userInfo></sipStack><autoAnswerTimer>1</autoAnswerTimer><autoAnswerAltBehavior>false</autoAnswerAltBehavior><autoAnswerOverride>true</autoAnswerOverride><transferOnhookEnabled>false</transferOnhookEnabled><enableVad>false</enableVad><preferredCodec>none</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>phoneLabel</phoneLabel><stutterMsgWaiting>0</stutterMsgWaiting><callStats>false</callStats><silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts><disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig><startMediaPort>10000</startMediaPort><stopMediaPort>20000</stopMediaPort><sipLines><line button="1"><featureID>9</featureID><featureLabel>Admin</featureLabel><proxy>sip. com</proxy><port>5060</port><name>111111</name><displayName>111111</displayName><autoAnswer><autoAnswerEnabled>2</autoAnswerEnabled></autoAnswer><callWaiting>3</callWaiting><authName>111111</authName><authPassword>SipPassword</authPassword><sharedLine>false</sharedLine><messageWaitingLampPolicy>1</messageWaitingLampPolicy><messagesNumber>80</messagesNumber><ringSettingIdle>4</ringSettingIdle><ringSettingActive>5</ringSettingActive><contact>105</contact><forwardCallInfoDisplay><callerName>true</callerName><callerNumber>true</callerNumber><redirectedNumber>false</redirectedNumber><dialedNumber>true</dialedNumber></forwardCallInfoDisplay></line><line button="2"><featureID>21</featureID><featureLabel>test</featureLabel><speedDialNumber>4444</speedDialNumber><featureOptionMask>1</featureOptionMask></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>SIP69xx. 9-4-1-3</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>1</webAccess><g722CodecSupport></g722CodecSupport><daysDisplayNotActive>1, 2, 3, 4, 5, 6, 7</daysDisplayNotActive><displayOnTime>08:30</displayOnTime><displayOnDuration>09:30</displayOnDuration><displayIdleTimeout>01:00</displayIdleTimeout><displayOnWhenIncomingCall>1</displayOnWhenIncomingCall><spanToPCPort>1</spanToPCPort><loggingDisplay>1</loggingDisplay><loadServer></loadServer></vendorConfig><networkLocale></networkLocale><networkLocaleInfo><name></name><version></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>
<processNodeName>pbx. com</processNodeName><proxy>pbx. com</proxy><name>1234-100</name><displayName>1234-100</displayName><authName>1234-100</authName>

Parametry w pliku konfiguracyjnym powinny zostać zmienione

<processNodeName>sip. com</processNodeName>; Wprowadź sip. com<proxy>sip. com</proxy>; Wprowadź sip. com<name>111111</name>; Numer SIP (na przykład 111111) z Panelu Klienta<displayName>111111</displayName>; Numer SIP (na przykład 111111) z Panelu Klienta<authName>111111</authName>; Numer SIP (na przykład 111111) z Panelu Klienta<authPassword>SipPassword</authPassword>; Hasło numeru SIP z Panelu Klienta<loadInformation>SIP69xx. 9-4-1-3</loadInformation>; Wersja oprogramowania wbudowanego. Należy ją określić w celu aktualizacji oprogramowania telefonu.<processNodeName>pbx. com</processNodeName>; Wprowadź pbx. com<proxy>pbx. com</proxy>; Wprowadź pbx. com<name>1234-100</name>; Twój numer wewnętrzny centrali telefonicznej z (na przykład 1234-100) z Panelu Klienta<displayName>1234-100</displayName>; Twój numer wewnętrzny centrali telefonicznej z (na przykład 1234-100) z Panelu Klienta<authName>1234-100</authName>; Twój numer wewnętrzny centrali telefonicznej z (na przykład 1234-100) z Panelu Klienta<authPassword>SipPassword</authPassword>; Twoje hasło numeru wewnętrznego centrali telefonicznej z Panelu Klienta

4. Telefon zaczyna pracę w trybie aktualizacji. Odłącz go od zasilania. Przyciśnij i przytrzymaj #, a następnie włącz telefon. Oba wskaźniki liniowe powinny świecić się na pomarańczowo. Na klawiaturze przyciśnij 123456789*0#. Jeżeli wszystko jest w porządku, telefon pobierze z serwera tftp wszystkie potrzebne pliki, a proces aktualizacji oprogramowania zacznie wyświetlać się na ekranie. W Tftpd32, w zakładce LOGS pokaże się informacja o pobranych plikach. Gdy aktualizacja oprogramowania zakończy się, telefon automatycznie uruchomi się ponownie i pobierze pliki konfiguracyjne. Jeżeli trzeba wprowadzić jakiekolwiek zmiany do pliku konfiguracyjnego, po edycji pliku załaduj go do telefonu – aby to zrobić, przyciśnij „Menu”, a następnie wybierz **#**. Telefon powinien automatycznie uruchomić się ponownie i pobrać tylko plik konfiguracyjny. Pliki zostaną pobrane przez telefon, gdy adres serwera tftp będzie znajdował się w ustawieniach. Jeżeli konfiguracja zakończy się pomyślnie, wejdź do menu telefonu i wyłącz serwer tftp Alternate tftp server > NO

Instrukcja demontażu i serwisowania Cisco Cisco 7960

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