Linksys SPA-901 IP Telephone

Linksys SPA-901 IP Telephone
    Linksys SPA-901 IP Telephone
  • Linksys SPA-901 IP Telephone

£49.99 (inc VAT)

£42.54 (exc VAT)

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€69.99 Add to cart

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Description

Linksys SPA-901 IP Telephone

Comprehensive Interoperability and SIP Based Feature Set

Based on the SIP Standard, the SPA901 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enablind service providers to quickly rollout competitive, feature-rich services to their customers.

Linksys SPA-901 Entry Level SIP VOIP Phone

With hudreds of features and configurable service parameters, the SPA-901 addresses the requirements of traditional business users while leveraging 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 SPA-901.

Carrier-Grade Security, Provisioning and Management

The SPA-901 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure 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 hassle and expense of managing, pre-loading and re-configuring customer premise equipment (CPE).

Warranty

1 Years Manufacturers Warranty

FAQs for ordinary use

Telephony Features

One service provider line
Two call appearances accessed via Flash Key or Hook Flash
Shared line appearance**
Line status indicator
Call Hold
Music on Hold**
Call Waiting
Outbound CallerID Blocking
Call transfer - Atended and Blind
Three Way conferencing with local mixing
Multi-Party Call Conferencing via external Conference Bridge**
Call Pick Up - Selective and Group**
Call park and UnPark**
Call back on Busy
Call Blocking - Anonymous and Selective
Call Forwarding - Unconditional, No Answer, On Busy
Call Return - Redial Last Caller
Hot Line and Warm Line Automatic Calling
Call Logs (60 Entries Each) Made, Answered and Missed Calls. Accessed via HTTP Server
Redial Last Called Number
Do Not Disturb (Caller Hears Busy Line tone)
Block Anonymous Incoming Calls
URI (IP) Dialing support (Vanity NUmbers)
Built-in Web Server for Administration and Configuration, with User and Admin Access Levels
Built-In Interactive Voice Response (IVR) System to check status and change configuration
Date and Time w/Intelligent Daylight Savings Support
Call Start Time stored in Call Logs
Distinctive Ringing
10 User-Downloadable Ring Tones - Ring Tone Generator free from http://www.linksys.com/
Speed Dial (8 entries)
Group Paging (Outbound Only)**
Intercom (Outbound Only)**
Set preferred CODEC, Per Call, All Calls
Configurable Dial/Numbering Plan Support
Ringer and Handset Voluem Controls
DNS SRV and Multiple A Records for Proxy Lookup and Proxy redundancy
Syslog, Debug, Report Generation, an Event Logging
Secure Call Encrypted Voice Communication Support
NAT Traversal
Automated Provisioning, Multiple Methods. Up to 256Bit encryption: (HTTP, HTTPS, TFTP)
Support Linksys Voice System Automatic Configuration
Optionally require Admin Password to Reset unit to factory defaults

**Feature requires support by call server.

Hardware

Voice Mail Message Waiting Indicator Light
Redial Button
Dedicated Flash Button
Volume Control button cycles through Voluem Levels. Controls Ringer and Handset Volume.
Standard 12-Button dialing pad
High Quality Handset and Cradle
Ethernet LAN - 10Base-T RJ-45
5v DC Universal (100-240v) Switching Power Adapter
 

Specifications:

Data Networking
MAC Address (IEEE 802.3)
IPv4
ARP
DNS
DHCP Client
ICMP
TCP
UDP
RTP
RTCP
DiffServ
VLAN Tagging
SNTP
Voice Gateway
SIPv2
SIP Proxy redundancy
Re-Registration with Primary SIP Proxy Server
SIP Support in NAT Networks (including STUN)
SIPFrag
Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
CODEC Name Assignment
G.711
G.726
G.729
G.723.1
Dynamic Payload Support
Adjustable Audio Frames Per Packet
DTMF: In-Band and Out-of-Band
Flexible Dial Plan Support with Inter-Digit Timers
I P Address / URI Dialing Support
Call Progress Tone Generation
Adaptive Jitter Buffer
Frame Los Concealment
VAD
Attenuation / Gain Adjustments
MWI and VMWI
Third Party Call Control

Security

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 AS Encryption