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Linksys SPA9000 SIP PBX

Linksys SPA9000 SIP PBX Appliance (1-16) Users




The SPA9000 marries the rich feature set of high-end PBX telephone systems with the convenience and cost advantages of Voice over IP. It has common voice system features such as an auto-attendant, shared line appearances, three way call conferencing, intercom, music on hold, call-forwarding and much more. The SPA9000 opens up access to the benefits of VoIP, including low cost long distance service, telephone number portability, and one network for both voice and data.




  • SIP Application Server, Proxy, Registrar and Location Server (RFC3261)
  • Multiple Service Provider Lines / SIP Account Support (4)
  • Shared Line Appearance (SLA)
  • Automated Attendant (AA)
  • Configurable AA Answer Delay
  • Interactive Voice Response (IVR)
  • Recordable IVR Prompts
  • Automatic Call Distribution (ACD)
  • Configurable Call Routing
    - Least Cost Routing
    - Multiple DID Numbers Per VoIP Line
    - Call Routing to Multiple Extensions or Targeted User
    - Call Hunting - Sequential, Round Robin, Random
  • Phone Configuration and Management Server
    - Discovery and Configuration of IP Phones
    - Assignment of Extension
    - Assignment of Dial plan
    - Proxy Logging of SIP Messages
    - Phone Firmware Upgrade Management
  • Corporate Directory with Automatic Update
  • Configuration and Maintenance via Web Interface (Local or Remote
    - Status Display of All Connections
  • Remote Configuration via
    - HTTPS with XML Formatted Files
    - HTTP or TFTP with 256-Bit Encrypted Binary Files
  • Call Park - User Definable Parking Space Number
  • Call Unpark
  • Call Transfer
  • Call Forward
  • Group Paging
  • Intercom
  • Directed Call Pick Up
  • Group Call Pick Up
  • Music / Information via Streaming Audio Server (SAS) for Calls:
    - On Hold
    - Parked in the Parking Lot
    - Being Transferred
  • Simultaneous Ringing (Find Me Service)
  • Do Not Disturb
  • Voice Mail Integration - Service Provider Based
    - Voice Mail Notification via SUBSCRIBE / NOTIFY
    - Forward Call Directly to Voice mail
  • Integrated Media Proxy or Direct RTP Routing to ITSP
  • Differentiated Services (DiffServ) / Type of Service (TOS) Support
  • Two FXS Ports for Phones, Fax machines, Media Adapters
  • Voice encoding according to G.711 (64kbit/s)
  • Fax Support using G.711 Pass-Through or T.38
  • Echo Cancellation (G.165)

Additional Features when used with SPA Phones

  • Line Status - Active Line Indication, Name/Number
  • Digits Dialed with Number Auto-Completion
  • Call Hold
  • Call Waiting
  • Call Transfer - Attended and Blind
  • Call Conferencing
  • Automatic Redial
  • Call Pick Up - Selective and Group **
  • Call Swap
  • Call Forwarding - Unconditional, No Answer, On Busy
  • Hot Line and Warm Line Automatic Calling
  • Call Log (60 entries each): Made, Answered, Missed Calls
  • Personal Directory with Auto-dial (100 entries)
  • Do Not Disturb
  • URI (IP) Dialing Support (Vanity Numbers)
  • On Hook Default Audio Configuration (Hands Free/Headset)
  • Multiple Ring Tones with Selectable Default Ring Tone per Line
  • Called Number with Directory Name Matching
  • Calling Number with Name - Directory Matching or via Caller ID
  • Subsequent Incoming Calls with Calling Name and Number
  • Date and Time with Intelligent Daylight Savings Support
  • Call Duration with Call Time Stamp Stored in Call Logs
  • Name/Identity (Text) Display at Start Up
  • Distinctive Ringing Based on Calling and Called Number
  • User Downloadable Ring Tones and Ring Tone Generator (Free from www.linksys.com)
  • Download on Demand Ring Tones - 10
  • Speed Dial Support
  • Configurable Dial/Numbering Plan Support - per Line
  • DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy
  • Syslog, Debug, Report Generation and Event Logging
  • Secure Call Encrypted Voice Communication Support
  • Built-in Web Server for Admin and Config with Multiple Security Levels
  • Automated Provisioning, Multiple Schemes-Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
  • Require Admin Password to Reset Unit to factory Defaults Option 

** Service feature availability is call feature server platform dependent,

Dimensions: 4.18 x 1.13. x 4.89in (106.17 x 28.7 x 124.2 mm) W x H x D
Unit Weight: 0.40 lbs (0.181 kg)

Data Networking

MAC Address (IEEE 802.3)
IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883)
ARP - Address Resolution Protocol
DNS - A Record (RFC 1706), SRV Record (RFC 2782)
DHCP Client - Dynamic Host Configuration Protocol (RFC 2131)
DHCP Server - Dynamic Host Configuration Protocol (RFC 2131)
PPoE Client - Point to Point Protocol over Ethernet (RFC 2516)
ICMP - Internet Control Message Protocol (RFC792)
TCP - Transmission Control Protocol (RFC793)
UDP - User Datagram Protocol (RFC768)
RTP - Real Time Protocol (RFC 1889) (RFC 1890)
RTCP - Real Time Control Protocol (RFC 1889)
DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349)
VLAN Tagging - 802.1p/q
SNTP - Simple Network Time Protocol (RFC 2030)
Upload Data Rate Limiting - Static and Automatic
QoS - Voice Packet Prioritization over Other Packet Types
Router or Bridge Mode of Operation
MAC Address Cloning
Port Forwarding

Voice Gateway

SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264)
SIP Proxy Redundancy - Dynamic via DNS SRV, A Records
Re-registration with Primary SIP Proxy Server
SIP Support in Network Address Translation Networks - NAT (incl. STUN)
Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
Codec Name Assignment
Voice Algorithms:
- G.711 (A-law and mμ-law)
- G.726 (16/24/32/40 kbps)
- G.729 A
- G.723.1 (6.3 kbps, 5.3 kbps)
Dynamic Payload Support
Adjustable Audio Frames Per Packet
DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO)
Flexible Dial Plan Support with Inter-Digit Timers
IP Address / URI Dialing Support
Call Progress Tone Generation
Jitter Buffer - Adaptive
Frame Loss Concealment
VAD - Voice Activity Detection w/ Silence Suppression
Attenuation / Gain Adjustments
MWI - Message Waiting Indicator Tones
Caller ID Support (Name & Number)

Provisioning, Administration & Maintenance:

Web Browser Administration & Configuration via Integral Web Server
Telephone Key Pad Configuration of Select Networking Parameters via IVR
Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP
Asynchronous Notification of Upgrade Availability via NOTIFY
Non-intrusive, In-Service Upgrades
Report Generation & Event Logging
Stats in BYE Message
Syslog & Debug Server Records - Per Line Configurable

Physical Interfaces:

2 10baseT RJ-45 Ethernet Port (IEEE 802.3) -- 1 WAN, 1 LAN
2 RJ-11 FXS Phone Ports - For Analog Circuit Telephone Device (Tip/Ring)

Advanced Multi-Port PSTN Gateway Solution
for the Linksys Voice System

The SPA400 features the ability to connect up to four (4) standard analog
telephones lines to a Linksys Voice System (LVS) VoIP network and includes the
additional benefit of an integrated voicemail application. The SPA400 utilizes
multiple analog phone lines and automatically routes calls to and from your
existing PSTN telephone service. Designed to be implemented with the LVS IP
Telephony System, the SPA400 enables cost-conscience business users to
leverage the high value features generally found only on more expensive PBX


Telephony Features:

Connects 4 Analog PSTN Lines to your VoIP network
4 RJ-11 FXO Ports
Integrated Voicemail Application Server
Enables SPA9000 IP Phone Clients to Leave and Playback Voicemail
USB 1.1 Host Interface for the Voicemail Module and Application
Configure up to 32 Voicemail Accounts
Caller ID Detection
Hunt Grouping of FXO Ports
Advanced Inbound and Outbound Call Routing
Independent Configurable Dial Plans to Utilize Multiple Service Providers
8 LEDs: Power, Status, LAN, Ports 1-4, USB
External Switching Power Adapter 5V DC/2A