Cisco SPA502G 1-Line IP Phone with Display, PoE and PC Port

Cisco SPA502G 1-Line IP Phone with Display, PoE and PC Port overview

Product 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

Product Overview

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)

Regulatory Compliance

FCC (Part 15, Class B), CE Mark, A-Tick, C-Tick, Telepermit, UL, CB

Security Features

Password-protected system, preset to factory default

Password-protected access to administrator and user-level features

HTTPS with factory-installed client certificate

HTTP digest: encrypted authentication via MD5 (RFC 1321)

Up to 256-bit Advanced Encryption Standard (AES) encryption

SIP over Transport Layer Security (TLS)

Secure Real-Time Transport Protocol (SRTP)

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)

Provisioning, administration, and maintenance

Integrated web server provides web-based administration and configuration

Telephone keypad configuration via display menu/navigation

Automated provisioning and upgrade via HTTPS, HTTP, TFTP

Asynchronous notification of upgrade availability via NOTIFY

Nonintrusive in-service upgrades

Report generation and event logging

Statistics transmitted in BYE message

Syslog and debug server records: configurable per line

Power supply

Power supply is optional and is purchased separately

Models: Cisco PA100-NA, PA100-EU, PA100-UK, PA100-AU

Switching type (100-240V) automatic

DC output voltage: +5 VDC at 2.0A maximum

Power adapter: 100-240V 50-60 Hz (26-34 VA) AC input

Physical interfaces

Two 10/100BASE-T RJ-45 Ethernet ports (IEEE 802.3)

Handset: RJ-9 connector

Built-in speakerphone and microphone

Headset 2.5mm jack

Indicator lights/LEDs

Speakerphone on/off button with LED

Headset on/off button with LED

Mute button with LED

Message waiting LED

Body dimensions (W x H x D)

8.42 x 8.35. x 1.73 in. (214 x 212 x 44 mm)

Unit weight

2.00 lb ( 0.9 kg)

Operating temperature

32º ~ 104ºF (0º ~ 40ºC)

Storage temperature

-4º ~ 158ºF (-20º ~ 70ºC)

Operating humidity

5% to 95% noncondensing

Storage humidity

5% to 95% noncondensing

Documentation

Quick-Start Installation and Configuration Guide

User Guide

Administration Guide

Provisioning Guide (for service providers only)

Package Contents

Cisco SPA502G 1-Line IP phone, handset, and stand

Handset cord

RJ-45 Ethernet cable

Quick-Start Installation

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Delivery Options

Next Working Day Delivery

Get your order delivered the next working day for £7.99, available when ordering Monday - Thursday and before 17:00 on that day if the product is in stock and your payment has cleared.

Pre 12 Next Working Day Delivery

Get your order delivered the next working day before 12:00pm for £10.98, available when ordering Monday - Thursday and before 17:00 on that day if the product is in stock and your payment has cleared. Some orders if via PayPal and the first customer order may not clear in time.

Weekend Delivery - Saturday & Sunday

If you need your order at the weekend, for £12.58, you can specify the day you wish to have it delivered to you, Saturday or Sunday providing your order has been placed by 17:00 on the Friday before.

Economy Delivery

Economy delivery is available on all orders of any value being shipped within the UK.