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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. Comprehensive Interoperability and SIP Based Feature Set Carrier-Grade Security, Provisioning, and Management
SPA962 Key Telephone Functions and Features • Up to Six Lines with Independent Configuration and Registration • 320 x 240 True Color, Four Inch, Liquid Crystal Display (LCD) • 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
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