FEATURES- 1-Line IP Telephone with 1 Ethernet Port and Hi-Res Display
       Advanced, Affordable, Feature Rich IP Phone for the Home Office and 
        Business.
       Stylish and functional in design, the SPA921 VoIP Phone is ideal for 
        a residence or business using a hosted IP telephony service, an IP 
        PBX, or a large scale IP Centrex deployment. The SPA921 leverages 
        industry leading VoIP technology from Linksys to deliver an 
        upgradeable high quality IP Phone that is unparalleled in features, 
        value, and support. 
       Standard features on the SPA921 include a high resolution graphical 
        display, speakerphone, and a 2.5 mm head-set port. The SPA921 
        supports one line with two call appearances and provides support 
        for three way conferencing, attended call transfer, and placing a 
        call on hold to answer an incoming call. The line can be configured 
        as a unique phone number (or extension), or can be configured to 
        share a number that is assigned to multiple phones. 
       Based on the SIP standard, the SPA921 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 sevice parameters, the SPA921 
        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 SPA921. 
       The SPA921 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 SPA921 IP Phone, Handset and Stand, 26" Handset Cord, 5v Power 
      Adapter with 6 ft. Cord, 6 ft. RJ45 Ethernet Cable, Quick 
      Installation Guide.
* Full featured one-line business class IP Phone 
* Connect directly to an Internet Telephone Service Provider or connect to 
  an IP PBX.
* Speakerphone. Caller ID. Call Hold, Transfer, Conferencing, and more.
* Easy installation with secure remote provisioning. Menu based and web 
  based configuration. 
     
  -- SPECIFICATIONs ------------------------------------
TELEPHONE FUNCTIONS-
               One Voice Line with Two Call Appearances
               Backlit Pixel Based Display: 128x64 Monochrome Graphical 
                 Liquid Crystal Display (LCD).
               Line Status - Active Line Indication, Name and Number
               Menu Driven User Interface
               Shared Line Appearance (requires support by call server)
               Speakerphone
               Call Hold
               Music on Hold (requires support by call server)
               Call Waiting
               Caller ID Name and Number and Outbound Caller ID Blocking
               Outbound Caller ID Blocking
               Call Transfer - Attended and Blind
               Three Way Call Conferencing with Local Mixing
               Connects to External Conference Bridge for Multi-party 
                 Conferencing.
               Automatic Redial of Last Calling and Last Called Numbers
                 On-Hook Dialing.
               Call Pick Up - Selective and Group (requires support by call 
                 server).
               Call Park and UnPark (requires support by call server)
               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)
               Digits Dialed with Number Auto-Completion
               Anonymous Caller Blocking
               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, Eight Entries
               Configurable Dial/Numbering Plan Support
               Intercom (requires support by call server)
               Group Paging (requires support by call server)
               NAT Traversal, including STUN Support
               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
               Built-in Web Server for Administration and Configuration 
                 with Multiple Security Levels.
               Automated Remote Provisioning, Multiple Methods. Up to 
                 256-Bit Encryption: (HTTP, HTTPS, TFTP).
               Optionally Require Admin Password to Reset Unit to Factory 
                 Defaults.
HARDWARE FEATURES-
               Pixel Based Display: 128x64 Monochrome LCD Graphical Display
               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
               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
               Ethernet LAN -- 10BaseT RJ-45
               5 volt DC Universal (100-240 Volt) Switching Power Adaptor
               LED Test Function
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 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)
SECURITY     - Password protected system reset to factory default 
               Password Protected Access to Administrator and User Level 
                 features.
               HTTPS with Factory Installed Client Certificate
               HTTP Digest - Encrypted Authentication via MD5 (RFC 1321)
               Up to 256-bit AES Encryption
PROVISIONING, ADMINISTRATION & MAINTENANCE-
               Integrated Web Server Provides Web Based Administration and 
                 Configuration.
               Integrated Voice Response system to report and modify 
                 configuration parameters.
               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-
               (1) 10BaseT/RJ45 port (IEEE 802.3)
               (1) Handset: RJ-7 connector
               (1) 2.5mm Headset port
INDICATORS   - Four (4) Call Appearance/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 - 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   - 7.68"w x 6.30"h x 7.09"d          WT.- 2.15 lbs.

Linksys 2-Line Home/Business VoIP Phone.