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Cisco SPA502G IP Phone

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Cisco SPA502G IP Phone

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Brand Cisco Systems

SKU SPA-502G

£77.90 exc. VAT

£93.48 inc. VAT

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10+	£75.00

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Cisco SPA 502G IP Phone

The Cisco SPA 502G is a 1-Line IP Phone with Display, PoE and PC Port that is part of the Cisco Small Business Pro Series. The SPA502G has been tested to ensure it is fully compatible with equipment from voice over IP (VoIP) infrastructure leaders. This enables service providers to quickly roll out feature-rich services to their customers.

Cisco SPA502G Key Features

Full-featured 1-line business-class IP phone supporting Power over Ethernet (PoE)
Supports up to two Cisco SPA500S Expansion Module, adding up to 64 additional buttons
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)
For business or home office use
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
Dependable, Affordable and Feature Rich
Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business

Carrier-Grade Security, Provisioning, and Management

The Cisco SPA 502G IP Phone 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.

Comprehensive Interoperability and SIP-Based Feature Set

One of the many advantages of the Cisco SPA 502G IP Phone is that it features easy station moves and shared line appearances (across local and geographically dispersed locations).

The Cisco SPA502G 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.

Note: Please refer to the Cisco VoIP Detailed Comparison Table for more information.

Cisco SPA 502G IP Phone Technical Specifications

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)
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)

Data Networking

MAC address (IEEE 802.3)
IPv4 (RFC 791)
Address Resolution Protocol (ARP)
DNS: A record (RFC 1706), SRV record (RFC 2782)
Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
Internet Control Message Protocol (ICMP) (RFC 792)
TCP (RFC 793)
User Datagram Protocol (UDP) (RFC 768)
Real-Time Transport Protocol (RTP) (RFC 1889, 1890)
Real-Time Control Protocol (RTCP) (RFC 1889)
Differentiated Services (DiffServ) (RFC 2475)
Type of service (ToS) (RFC 791, 1349)
VLAN tagging 802.1p/Q: Layer 2 quality of service (QoS)
Simple Network Time Protocol (SNTP) (RFC 2030)

Voice Gateway

SIP version 2 (RFC 3261, 3262, 3263, 3264)
SPCP with the Cisco Unified Communications 500 Series
SIP proxy redundancy: dynamic via DNS SRV, A records
Reregistration with primary SIP proxy server
SIP support in NAT networks (including STUN)
SIPFrag (RFC 3420)
Secure (encrypted) calling via SRTP
Codec name assignment
Voice algorithms: G.711 (A-law and µ-law), G.726 (16/24/32/40 kbps), G.729 A, G.722
Dynamic payload support
Adjustable audio frames per packet
Dual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO)
Flexible dial plan support with interdigit timers
IP address/URI dialing support
Call progress tone generation
Jitter buffer: adaptive
Frame loss concealment
Comfort Noise Generation (CNG)
Voice activity detection (VAD) with silence suppression
Attenuation/gain adjustments
VMWI - Voicemail Waiting Indicator, via NOTIFY, SUBSCRIBE
Caller ID support (name and number)
Third-party call control (RFC 3725)