Cisco SPA962 User Manual

• 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 

Model No.