SIP922 : 1-Line IP Telephone with 2-Port Ethernet Switch and Hi-Res Display Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business

Stylish and functional in design, the SPA922 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 SPA922 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP Phone that is unparalleled in features, value, and support.

• 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 and secure remote provisioning. Menu based and web based configuration.
• An optional power supply is orderable separately for this product if not utilizing Power over Ethernet(PoE)

Standard features on the SPA922 include dual switched Ethernet ports, 802.3af PoE, a high resolution graphical display, speakerphone and a 2.5 mm head-set port. The SPA922 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.

• 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