The Cisco SPA508G is part of the Cisco Small Business Pro Series. The SIP-based Cisco SPA508G 8-Line IP Phone has been tested to ensure comprehensive interoperability with equipment from voice over IP (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 Cisco SPA508G 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 SPA508G.
The Cisco SPA508G 8-Line IP Phone also supports productivity-enhancing features such as VoiceView Express, and Cisco XML applications when used with the Cisco Unified Communications 500 Series in SPCP mode.
**Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business**
Telephony Features
Eight Voice Lines
Eight independent SIP Registrations*
Line status: active line indication, with 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 transfer
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 bus
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
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 start time stored in call logs
Call timer
Name and identity (text) displayed at startup
Distinctive ringing based on calling and called number
10 user-downloadable ring tones
Speed dialing, eight entries
Configurable dial/numbering plan support
intercom**
Group paging**
Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support
DNS SRV and multiple A records for proxy lookup and proxy redundancy
Syslog, debug, report generation, and event logging
Highly secure call encrypted voice communications 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, Trivial File Transfer Protocol [TFTP])
Option to require administrator password to reset unit to factory defaults
*Feature supported only in SIP mode
**Feature requires support by call server
Hardware Features
Pixel-based display: 128 x 64 monochrome LCD graphical display with backlight
All information is provided for reference only. If you are unsure about any of the features listed, please check the manufacturer's official information.