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6LINE IP TELEPHONE WITH 2PORT SWITCH POE & DISPLAY

6Line Ip Telephone With 2Port Switch Poe & Display
Price: US$255.84
In Stock. Usually ships in 1-2 business days
Platform: Windows
Category: Windows Telephones
By: Cisco Systems - Cobo
Manufacturer Part#: SPA962-NA
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FEATURES- 6-Line IP Phone with Color Display
       6-Line IP Phone with Color Display Enhances Business Communication 
         Requirements!
       Stylish and functional in design, the SPA962 VoIP telephone is a 
        must for businesses using a hosted IP telephony service, an IP PBX, 
        or a large scale IP Centrex deployment. The SPA962 leverages 
        industry leading VoIP technology from Linksys to deliver a high 
        quality IP Phone that is unparalleled in features, value, and 
        support. 
       Standard features on the SPA962 include six active lines, dual 
        switched Ethernet ports, 802.3af PoE support, a high resolution 
        color display, speakerphone, and a 2.5 mm head-set port. Each line 
        can be independently configured to use a unique phone number (or 
        extension), or can be configured to use a shared number that is 
        assigned to multiple phones. The power supply for the SPA962 is 
        sold separately and will be required if PoE functionality is not 
        implemented. 
       Based on the SIP standard, the SPA962 has been tested to ensure 
        comprehensive interoperability with equipment from VoIP 
        infrastructure leaders enabling service providers to quickly roll-
        out competitive, feature rich services to their customers. With 
        hundreds of features and configurable service parameters, the 
        SPA962 addresses the requirements of traditional business users 
        while leveraging the advantages of IP telephony. Features such as 
        easy station moves, presence, and shared line appearances (across 
        local and geographically dispersed locations) are just some of the 
        many advantages of the SPA962. 
       The SPA962 uses standard encryption protocols to provide secure 
        remote provisioning and unobtrusive in-service software upgrades. 
        Linksys secure remote provisioning tools include detailed 
        performance measurement and troubleshooting features, enabling 
        network providers to deliver high quality support to their 
        subscribers. Remote provisioning also saves service providers the 
        hassle and expense of managing, pre-loading, and re-configuring 
        customer premise equipment (CPE). 
    Includes SPA962 IP Phone, Handset, and Stand; Handset Cord; RJ45 
      Ethernet Cable; Quick Installation Guide. Power Supply is Ordered 
      Separately.
* Full featured six line business class IP Phone supporting Power over 
  Ethernet 802.3af.
* Connect directly to an Internet Telephone Service Provider or connect to 
  an IP PBX.
* Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, 
  Conferencing, and more.
* Appealing Four Inch, True Color Liquid Crystal Display (LCD).
     
  -- SPECIFICATIONs ------------------------------------
TELEPHONE FUNCTIONS-
               Up to Six Lines with Independent Configuration and 
                 Registration.
               Secure Call Support - SIP over TLS, and SRTP
               Line Status - Active Line Indication, Name and Number
               Menu Driven User Interface - Multiple Languages Supported
               Digits Dialed with Number Auto-Completion
               Shared/Bridged Line Appearance **
               High Quality Speakerphone
               Call Hold
               Music on Hold **
               Call Waiting
               Caller ID Name and Number and Outbound Caller ID Blocking
               Outbound Caller ID Blocking
               Call Transfer - Attended and Blind
               Call Conferencing
               Automatic Redial
               On-Hook Dialing
               Call Pick Up - Selective and Group **
               Call Park and UnPark **
               Call Swap
               Call Back on Busy
               Call Blocking - Anonymous and Selective
               Call Forwarding - Unconditional, No Answer, On Busy
               Hot Line and Warm Line Automatic Calling
               Call Logs (60 entries each): Made, Answered, and Missed 
                 Calls.
               Redial from Call Logs
               Personal Directory with Auto-dial (100 entries)
               Do Not Disturb (callers hear line busy tone)
               URI (IP) Dialing Support (Vanity Numbers)
               On Hook Default Audio Configuration (Speakerphone and 
                 Headset).
               Multiple Ring Tones with Selectable Ring Tone per Line
               Called Number with Directory Name Matching
               Call Number using Name - Directory Matching or via Caller ID
               Subsequent Incoming Calls with Calling Name and Number
               Date and Time with Intelligent Daylight Savings Support
               Call Duration and Start Time Stored in Call Logs
               Call Timer
               Name and Identity (Text) Displayed at Start Up
               Distinctive Ringing Based on Calling and Called Number
               Ten User Downloadable Ring Tones - Ring Tone Generator Free 
                 from www.linksys.com
               Speed Dialing
               Configurable Dial/Numbering Plan Support - per Line
               Intercom **
               Group Paging **
               DNS SRV and Multiple A Records for Proxy Lookup and Proxy 
                 Redundancy.
               Syslog, Debug, Report Generation, and Event Logging
               Secure Call Encrypted Voice Communication Support - SIP over 
                 TLS, and SRTP.
               Built-in Web Server for Administration and Configuration 
                 with Multiple Security Levels.
               Automated Provisioning, Multiple Methods. Up to 256-bit 
                 Encryption: (HTTP, HTTPS, TFTP).
               Optionally Require Admin Password to Reset Unit to factory 
                 Defaults.
           ** Feature requires support by SIP server
HARDWARE     - 320 x 240 True Color, 4" Liquid Crystal Display (LCD)
               4 Illuminated Call Appearance Line Buttons w/Tricolor LED's
               LED Indicates Line State - Active, Idle, On-Hold, 
                 Unregistered.
               Line LED Configurable to 13 Different States (On/Off, Color, 
                 Flash).
               Dedicated Illuminated Buttons for: Audio Mute On/Off
                                                  Headset On/Off
                                                  Speakerphone On/Off
               Four Soft Key Buttons
               Four Way Rocking Directional Knob for Menu Navigation
               Support for up to two attached Attendant Consoles; adds up 
                 to 64 programmable buttons.
               Voice Mail Message Waiting Indicator Light
               Voice Mail Message Retrieval Button
               Dedicated Hold Button
               Settings Button for Access to Feature, Set-up, and 
                 Configuration Menus.
               Volume Control Rocking Up/Down Knob Controls Handset, 
                 Headset, Speaker, Ringer.
               Standard 12-Button Dialing Pad
               High Quality Handset and Cradle
               Built-In High Quality Microphone and Speaker
               Headset Jack: 2.5 millimeter
               LED Test Function
               Two Ethernet LAN Ports with Integrated Ethernet Switch - 
                 100BaseT RJ-45.
               802.3af Compliant Power over Ethernet (PoE)
               Optional 5 volt DC Universal (100-240 Volt) Switching - 
                 Power Supply is Ordered Separately.
DATA NETWORKING-
               MAC Address (IEEEE-802.3)
               IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 
                 1883).
               ARP - Address Resolution Protocol
               DNS - A Record (RFC 1706), SRV Record (RFC 2782)
               DHCP Client - Dynamic Host Configuration Protocol (RFC 2131)
               ICMP - Internet Control Message Protocol (RFC 792)
               TCP - Transmission Control Protocol (RFC 793)
               UDP - User Datagram Protocol (RFC 768)
               RTP - Real Time Protocol (RFC 1889)(RFC 1890)
               RTCP - Real Time Control Protocol (RFC 1889)
               DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349)
               VLAN Tagging 802.1p/q - Layer 2 QoS
               SNTP - Simple Network Time Protocol (RFC 2030)
VOICE GATEWAY- SIPv2 - Session Initiation Protocol v2 (RFC 3261, 3262, 
                 3263, 3264).
               SIP over TLS
               SRTP
               Secure (Encrypted) Calling via Pre-Standard Implementation 
                 of Secure RTP.
               SIP Proxy Redundancy - Dynamic via DNS SRV, A Records
               Re-registration with Primary SIP Proxy Server
               SIP Support in Network Address Translation Network - NAT 
                 (including STUN).
               SIPFrag (RFC 3420)
               Secure (Encyrpted) Calling via Pre-Standard implementation 
                 of Secure RTP.
               Codec Name Assignment
               Voice Algorithms: G.711 (A-law and mu-law)
                                 G.726 (16/24/32/40 kbps)
                                 G.729A
                                 G.723.1 (6.3 kbps, 5.3 kbps)
               Dynamic Payload Support
               Adjustable audio frames per packet
               DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO)
               Flexible Dial Plan Support with Configurable InterDigit 
                 Timers.
               IP Address/URI Dialing Support
               Call Progress Tone Generation
               Jitter Buffer - Adaptive
               Frame Loss Concealment
               VAD - Voice Activity Detection with Silence Suppression
               Attentuation/Cain/Adjustments
               MWI- Message Waiting Indicator tones
               VMWI - Visual Message Waiting Indicator - Via NOTIFY, 
                 SUBSCRIBE.
               Third Party Call Control (RFC 3725)
PROVISIONING, ADMINISTRATION & MAINTENANCE-
               Integrated Web Server Provides Web Based Administration and 
                 Configuration.
               Telephone Key Pad Configuration via Display Menu/Navigation
               Automated Provisioning and Upgrade via HTTPS, HTTP, TFTP
               Asynchronous Notification of Upgrade Availability via NOTIFY
               Non-intrusive, In-Service Upgrades
               Report Generation and Event Logging
               Statistics Transmitted in BYE Message
               Syslog and Debug Server Records - Configurable Per Line
DATA INTERFACES-
               (2) 100BaseTX/RJ45 ports (IEEE 802.3, including 802.3af)
               (1) Handset: RJ-7 connector
               (1) 2.5mm Headset port
               Built-in Speakerphone and Microphone
INDICATORS   - (6) Line Buttons with Associated Tricolor LED
               Line LED State Indication: Active, Idle, On Hold, 
                 Unregistered.
               Speakerphone On/Off Button with LED
               Headset On/Off Button with LED
               Mute Button with LED
               Message Waiting Indicator LED
               Voicemail Message Retrieval Button
               Hold Button
               LED Test Function
APPROVALS    - FCC B, CE, A-Tick
POWER SUPPLY - Power supply is optional and is purchased separately
                 -- Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU
                 Switching Type (100-240v) Automatic.
               DC Input Voltage : +5 Volts DC at 2.0 Amps Maximum
               Power Consumption: 5 Watts
               Power Adapter    : 100-240v, 50/60Hz (26-34VA) AC Input
DIMENSIONS   - 8"w x 7.63"h x 7.50"d          WT.- 2.4 lbs.

The Linksys SPA962-NA 6-Line IP Phone with Color Display.

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