Cisco SPA512G Summary

The Cisco SPA512G 1-Line IP Phone with 2-Port Gigabit Ethernet Switch provides advanced voice and data communications features small businesses need to stay productive and responsive. Based on Session Initiation Protocol (SIP), the Cisco SPA512G has been tested to ensure complete interoperability with leading voice over IP (VoIP) equipment from voice over IP 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 Cisco SPA512G addresses the requirements of traditional business users while building on the advantages of IP telephony. Features such as easy station moves and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA512G.

Stylish and affordable, the Cisco SPA512G IP Phone can also be used with productivity-enhancing features such as VoiceView Express, and Cisco XML Applications when interfacing with Cisco Unified Communication 500 Series in SPCP Mode.

• Full-featured 1-line business-class IP phone supporting Power over Ethernet (PoE)
• Connects directly to an Internet telephone service provider or to an IP PBX
• Dual Gigabit Ethernet switched ports, speakerphone, caller ID, call hold, conferencing, and more
• Easy installation and secure remote provisioning, as well as menu-based and web-based configuration
• Supports up to two Cisco SPA500S Expansion Modules, adding up to 64 additional buttons
• Supports both SIP and SPCP Protocols with Cisco Unified Communication 500 Series

Cisco SPA512G Core Features

• Four voice lines
• Backlit pixel-based display: 128 x 64 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
• Outbound caller ID blocking
• Call transfer: attended and blind
• Three-way call conferencing with local mixing
• Multiparty conferencing via external conference bridge
• Automatic redial of last calling and last called numbers
• On-hook dialing
• Call pickup: 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
• Uniform Resource Identifier (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
• Ability to call number using name: directory matching or via caller ID
• Subsequent incoming calls show calling name and number
• Date and time with support for intelligent daylight savings
• Call duration and start time stored in call logs
• Call timer
• Name and identity (text) displayed at startup
• Distinctive ringing based on calling and called number
• 12 user-customizable ring tones
• 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
• Debug, syslog, 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 secure provisioning via TFTP, HTTP or HTTPS
• Option to require administrator password to reset unit to factory defaults