Condition: New

Cisco Part Numbers: SPA922-EU

• Full featured one-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
  
• Easy installation with secure remote provisioning. Menu based and web based configuration.

Comprehensive Interoperability and SIP Based Feature Set

Based on the SIP standard, the SPA-922 has been tested to ensure comprehensive interoperability with equipment enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable sevice parameters, the SPA922 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 SPA-922.

Carrier-Grade Security, Provisioning, and Management

The SPA922 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, preloading, and re-configuring customer premise equipment (CPE).

Package Contents

• 1 - SPA-922 IP Phone, Handset, and Stand
• 1 - Handset Cord
• 1 - RJ45 Ethernet Cable
• 1 - Quick Installation Guide

• 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 **
• 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
• 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 **
• 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)
• 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 **
• Group Paging **
• 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

** Feature requires support by call server

Hardware Features

• Pixel Based Display: 128x64 Monochrome LCD Graphical Display with back light
• Dedicated Illuminated Buttons for:
  • Audio Mute On/Off
  • Headset On/Off
  • Speakerphone On/Off
• 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
• 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

• MAC Address (IEEE 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 (RFC792)
• TCP - Transmission Control Protocol (RFC793)
• UDP - User Datagram Protocol (RFC768)
• 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)

• SIPv2 - Session Initiation Protocol Version 2 (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 Networks - NAT (including STUN)
• SIPFrag (RFC 3420)
• Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
• Codec Name Assignment
• Voice Algorithms:
  • - G.711 (A-law and μ-law)
  • - G.726 (16/24/32/40 kbps)
  • - G.729 A
  • - G.723.1 (6.3 kbps, 5.3 kbps)
• Dynamic Payload Support
• Adjustable Audio Frames Per Packet
• DTMF: In-band and Out-of-Band (RFC 2833) (SIP INFO)
• Flexible Dial Plan Support with Inter-Digit Timers
• IP Address / URI Dialing Support
• Call Progress Tone Generation
• Jitter Buffer - Adaptive
• Frame Loss Concealment
• VAD - Voice Activity Detection with Silence Suppression
• Attenuation / Gain Adjustments
• MWI - Message Waiting Indicator Tones
• VMWI - Voice Mail Waiting Indicator - Via NOTIFY, SUBSCRIBE
• Caller ID Support (Name and Number)
• 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

• 2 100baseT RJ-45 Ethernet Port (IEEE 802.3)
• Handset: RJ-7 Connector
• Built-in Speakerphone and Microphone
• Headset 2.5 mm Port
  
   

• Power suppy 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

• 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