SPA962 IP Phone

6-Line IP Phone with Color Display Enhances Business Communication Requirements

Stylish and functional in design, the SPA962VolP 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.

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.

Standard features on the SPA962 include six active lines, dual-switched Ethernet ports, 802.3af Power over Ethernet (PoE)* support, a high-resolution color display, full-duplex speakerphone, and a 2.5 mm headset 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 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, preloading, and re-configuring customer premise equipment (CPE).

*The power supply for the SPA962 is sold separately and is required if PoE functionality is not implemented.

  • Full-featured six-line business-class IP Phone supporting PoE 802.3af
  • Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
  • Dual-switched Ethernet ports, full-duplex speakerphone, caller ID, call hold, conferencing, and more
  • Appealing four-inch, true color Liquid Crystal Display (LCD)
Specifications
Hardware
  • 320 x 240 True Color, Four Inch, Liquid Crystal Display (LCD)
  • Six Illuminated Call Appearance Line Buttons with Tricolor LEDs
  • LED Indicates Line StateActive, Idle, On-Hold, Unregistered
  • Line LED Configurable to 13 Different States (On/Off, Color, Flash)
  • Dedicated Illuminated On/Off Buttons for Audio Mute, Headset, and Speakerphone
  • Four Soft Key Buttons
  • Four Way-Rocking Directional Button 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 Button Controls Handset, Headset, Speaker, Ringer
  • Standard 12-Button Dialing Pad
  • High-Quality Handset (RJ-7 connector) and Cradle
  • Built-In High Quality Microphone and Full-Duplex Speakerphone
  • Headset Jack2.5 millimeter port
  • 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)
Security
  • Password-Protected System, Preset to Factory Default
  • Password-Protected Access to Administrator and User Level Features
  • HTTPS with Factory-Installed Client Certificate
  • HTTP DigestEncrypted Authentication via MD5 (RFC 1321)
  • Up to 256-bit AES Encryption
Documentation
  • Installation and Configuration Guide
  • User Guide
  • Administration Guide
  • Provisioning GuideFor Service Providers Only
Environmental
Dimensions 203 x 194 x 191 mm
Weight 1.088 g
Power
  • DC Input Voltage: +5 Volts DC at 1.0 Amps Maximum
  • Power Consumption: 5 Watts (tentative)
  • Switching Type (100-240v) Automatic
  • Optional Power Adapter (models PA100-NA, PA100-EU, PA100-UK,
  • PA100-AU): 100-240v - 50-60Hz (26-34VA) AC Input
Certification FCC, CE, Class B Canadian ICES-003, A-Tick
Operating Temp. 41 to 113F (5 to 45C)
Storage Temp. -13 to 185F (-25 to 85C)
Operating Humidity 10 to 90%, Noncondensing
Storage Humidity 10 to 90%, Noncondensing
Package Contents
  • 1 SPA962 IP Phone, Handset, and Stand
  • 1 Handset Cord
  • 1 RJ-45 Ethernet Cable
  • 1 Quick Installation Guide
  • (Optional Power Supply is Ordered Separately)
Functions and Features
  • Up to Six Lines with Independent Configuration and Registration
  • Secure Call SupportSIP over TLS, and SRTP
  • Line StatusActive Line Indication, Name and Number
  • Menu-Driven User InterfaceMultiple Languages Supported
  • Digits Dialed with Number Auto-Completion
  • Shared/Bridged Line Appearance**
  • High-Quality Full-Duplex Speakerphone
  • Call Hold
  • Music on Hold**
  • Call Waiting
  • Caller ID Name and Number
  • Outbound Caller ID Blocking
  • Call TransferAttended and Blind
  • Call Conferencing
  • Automatic Redial
  • On-Hook Dialing
  • Call Pick UpSelective and Group**
  • Call Park and Retrieval**
  • Call Swap
  • Call Back on Busy
  • Call BlockingAnonymous and Selective
  • Call ForwardingUnconditional, 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 NameDirectory 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 TonesFree Ring Tone Generator 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 and Debug Server Records (Configurable Per Line)
  • Report Generation and Event Logging
  • Statistics Transmitted in BYE Message
  • 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 Methodsup to 256 Bit Encryption (HTTP, HTTPS, TFTP)
  • Asynchronous Notification of Upgrade Availability via NOTIFY
  • Non-intrusive, In-Service Upgrades
  • Optionally Require Admin Password to Reset Unit to factory Defaults
  • ** Feature requires support by SIP server
Data Networking
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)
Voice
  • SIPv2Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264)
  • SIP over TLS
  • SRTP
  • SIP Proxy RedundancyDynamic via DNS SRV, A Records
  • Re-registration with Primary SIP Proxy Server
  • SIP Support in Network Address Translation NetworksNAT (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
  • DTMFIn-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 BufferAdaptive
  • Frame Loss Concealment
  • VADVoice Activity Detection with Silence Suppression
  • Attenuation/Gain Adjustments
  • MWIMessage Waiting Indicator Tones
  • VMWIVoice Mail Waiting IndicatorVia NOTIFY, SUBSCRIBE
  • Caller ID Support (Name and Number)
  • Third Party Call Control (RFC 3725)

Linksys IP Telephone Comparison Chart

SPA Model Voice Lines Ethernet Ports High Resolution Graphical Display Power over Ethernet Support
SPA901 1 1 N N
SPA921 1 1 Y N
SPA922 1 2 Y Y
SPA941 2-4 1 Y N
SPA942 2-4 2 Y Y
SPA962 6 2 Yes, Color Yes