Cisco SPA502G

Highlights

For business or home office use
Dependable, Affordable and Feature Rich
Full-featured 1-line business-class IP phone supporting Power over Ethernet (PoE)
Monochrome backlit display for ease of use, aesthetics, and on-screen applications
Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
Dual switched Ethernet ports for connecting a computer behind the phone, reducing cabling costs
Wideband audio for unsurpassed voice clarity and enhanced speaker quality
Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration
Supports up to two Cisco® SPA500S Expansion Module, adding up to 64 additional buttons*
Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco
Unified Communications 500 Series for Small Business

Comprehensive Interoperability and SIP-Based Feature Set
Part of the Cisco Small Business Pro Series, the SIP based Cisco SPA502G 1-Line IP Phone (Figure 1) 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 SPA502G 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 SPA502G.

The Cisco SPA502G 1-Line IP phone also supports productivity-enhancing features such as VoiceView Express and Cisco XML applications when used with Cisco Unified Communications 500 Series in SPCP mode.

Carrier-Grade Security, Provisioning, and Management
The Cisco SPA502G uses standard encryption protocols to perform highly secure remote provisioning and unobtrusive in-service software upgrades. Remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high-quality support to their subscribers. Remote provisioning also saves service providers the time and expense of managing, preloading, and reconfiguring customer premises equipment.

Telephony Features

One voice line
One SIP registration
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
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
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
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

Hardware Features

Pixel-based display: 128 x 64 monochrome LCD graphical display with backlight
Dedicated illuminated buttons for:
- Audio mute on/off
- Headset on/off
- Speakerphone on/off
4-way rocking directional knob for menu navigation
Voicemail message waiting indicator (VMWI) light
Voicemail message retrieval button
Dedicated hold button
Settings button for access to feature, setup, and configuration menus
Volume control rocking up/down knob controls handset, headset, speaker, ringer
Standard 12-button dialing pad
High-quality handset and cradle
Built-in high-quality microphone and speaker
Headset jack: 2.5 mm
LED test function
Two Ethernet ports with integrated Ethernet switch: 10/100BASE-T RJ-45
802.3af-compliant PoE
Optional 5 VDC universal (100-240V) switching; power supply is ordered separately (Cisco PA100)