Linksys Sipura SPA3102 VoIP Phone adapter.

Features 1 x WAN port, 1 x LAN port, 1 x FXO port and 1 X FXS port. 

This is NOT a locked unit.

Connect a regular phone or fax machine to the SPA3000 and dramatically cut your call costs by using VoIP technology with your broadband internet connection.

This unit can also be used with an IP PBX system like the Linksys SPA9000 to provide PSTN Failback or convert a standard phone or fax machine to an IP telephony device.

The SPA3102 will automatically fall back to your regular PSTN phone line if your ISP, ITSP or hardware infrastructure fails, providing you with a seamless reliable communications service.

It is compatible with all VoIP (SIP) service providers.

SPA3102  $117.00 Inc GST

       SPA3102 Specifications

• Automatically routing of local calls from a mobile.
• Phone and landline to VOIP service.
• Reduce international & long distance phone charges.
• Calls can be sent via a traditional carrier via the PSTN interface.
• PSTN Interface - good in power outs or if the internet is down
• Clear, high-quality voice communication in changeable IP network environments.
• The SPA3102 provides market leading, best-in-class VoIP endpoint functionality with the added benefit of an integral connection for legacy telephone networks

The SPA3102 features the ability to connect standard telephones and fax machines to IP-based data networks with the additional benefit of an integrated connection for legacy telephone network "hop-on, hop-off" applications. SPA3102 users will be able to leverage their broadband phone service more than ever by automatically routing local calls from mobile phones and land lines over to VoIP service providers and vice versa. If power is lost to the unit or Internet service is down, calls can be redirected to a traditional carrier via the FXO interface.

A user calling from a mobile phone or land line will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA3102 via a local telephone number. The advanced authentication and call routing intelligence programmed into the SPA3102 will route the call via the Internet to the far end destination. In addition, when using the SPA3102 at the far end, VoIP calls placed to that location can be either answered or further processed and routed on as a local call to any legacy land line or mobile phone.

The SPA3102 supports one RJ-11 POTS (Plain Old Telephone Service) FXS port to connect an existing analog phone or fax machine. The SPA3102 also supports one PSTN FXO port to connect to a Telco or PBX circuit. The SPA3102 includes 2 100BaseT RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to a broadband modem or router. The SPA3102 FXS and FXO lines can be independently configured via software controlled by the service provider or the end user.

Installed by the end user and remotely provisioned, configured and maintained by the service provider, each SPA3102 converts voice traffic into data packets for transmission over an IP network. Compact in design, the SPA3102 can be used in consumer and business VoIP service offerings including a full-featured IP Centrex environment. The SPA3102 uses international standards for voice and data networking for reliable voice and fax operation.

Toll Quality Voice and Carrier-Grade Feature Support
The SPA3102 delivers clear, high-quality voice communication in diverse network conditions. Excellent voice quality in a demanding IP network is consistently achieved via our advanced implementation of standard voice coding algorithms. The SPA3102 is interoperable with common telephony equipment like voicemail, Fax, PBX, and interactive voice response systems.

Large-Scale Deployment and Management
The SPA3102 offers all the key features and capabilities which service providers can provide customized VoIP services to their subscribers. The SPA3102 can be remotely provisioned and supports dynamic, in-service software upgrades. A secure profile upload saves providers the time, expense, and hassle of managing and pre-configuring or re-configuring customer premise equipment (CPE) for deployment.
Ironclad Security

Linksys understands that security for end users and service providers is a fundamental requirement for a solid, carrier-grade telephony service. The SPA3102 supports secure, standard encryption-based methods for communication, provisioning and servicing.

      Telephony

• Service Authentication via PIN, Digest, Caller ID                        (Bellcore Type 1)

• Per Call Authentication and Associated Routing

• Least Cost Routing Support

• Impedance Agnostics - 8 Settings

• Call Waiting, Cancel Call Waiting, Call Waiting Caller ID Detection (Bellcore Type 1)

• Caller ID with Name/Number (Multi-national Variants)

• Caller ID Blocking

• Call Forwarding to PSTN or VoIP Service: No answer, Busy, All

• Do Not Disturb

• Call Transfer

• Three-way Conference Calling with Local Mixing

• Message Waiting Indication - Visual and Tone Based

• Call Return

• Call Back on Busy

• Call Blocking with Toll Restriction

• Delayed Disconnect

• Distinctive Ringing - Calling and Called Number

• Off-hook Warning Tone

• Selective/Anonymous Call Rejection

• Hot line and Warm Line Calling

• Speed Dialling of 8 Numbers/Addresses

• Music on Hold

• Fax: G.711 Pass Through or Real Time Fax over IP via T.38

      Product Specific (SPA3102)

• VoIP to PSTN (USA) Service Call Origination and Termination

• PSTN (USA) to VoIP Service Call Origination and Termination

• Single Stage and Two Stage Dialling

• Forward Calls to VoIP service - Selective, Authenticated, All

• Forward Calls to PSTN service - Selective, Authenticated, All

• PSTN Line Sharing with Multiple Extension

• Automatic PSTN Fallback (Loss of Power or IP Service to Unit - with Quiescence to Normal Operations)

• Advanced Inbound and Outbound Call Routing

• Independent Configurable Dial Plans - Up to 8

• Force PSTN Disconnection

• Sequential Dialling Support

      VoIP to PSTN Authentication and Routing Features

• VoIP to PSTN Gateway Enable/Disable

• VoIP Caller Auth Method (None, PIN, HTTP Digest)

• VoIP PIN Max Retry Setting

• One Stage Dialling Enable/Disable

• VoIP Caller ID Pattern Matching

• VoIP Access Allowed Caller List (No Further Authentication)

• VoIP Caller PIN and Associated Dial Plan

      PSTN to VoIP Authentication and Features

• PSTN to VoIP Gateway Enable/Disable

• VoIP Caller Auth Method (None, PIN, HTTP Digest)

• Ring Through to FXS Enable/Disable

• Ring Through Tone - Configurable

• Caller ID (Bellcore Type 1) for VoIP Service Acces

• Caller ID Enable/Disable

• PIN Max Retry Settings

• Access Allowed Caller List (No Further Authentication)

• Caller PIN and Associated Dial Plan

• Least Cost Routing (via Outbound VoIP - Line1 Dial Plan)

      FXO Behavior Features

• VoIP Answer Delay Timer

• PSTN Answer Delay Timer

• VoIP PIN Digit Time-Out Timer

• PSTN PIN Digit Time-Out Timer

• PSTN-to-VoIP Call Max Dur Timer

• VoIP-to PSTN Call Max Dur Timer

• PSTN Ring Through Delay Timer

• PSTN Dialing Delay Timer

• VoIP DIG Refresh Interval Timer

• PSTN Ring Time-out Timer

      PSTN Disconnection Detection Features

• CPC (Removal of Tip/Ring Voltage Momentarily)

• Polarity Reversal

• Long Silence (Configurable Time Setting)

• Disconnect Tone (e.g. Reorder Tone)

• Silence Threshold

      International Control Features

• FXO Port Impedance - Configurable to 16 settings

• Ring Frequency - Configurable

• SPA to PSTN and PSTN to SPA Gain Settings

• Ring Frequency - Maximum Setting

• Ring Validation Time Setting

• Tip/Ring Voltage Adjustment Setting

• Ring Indication Delay Setting

• Operational Loop Current Minimum Value

• Ring Time-out Setting

• On-Hook Speed Setting

• Ringer Impedance Setting

• Line-in-Use Voltage Setting

• 1 - SPA3102 Phone Adapter Unit

• 1 - Power Adapter

• 1 - RJ-45 Ethernet Cable

• 1 - RJ-11 Telephone Cable

• 1- Quick Installation Guide

Model SPA3102                                                                        Many specifications are programmable within a defined range or list of options. Please see the SPA Administration Guide for details. The configuration profile is uploaded to the SPA3102 at the time of provisioning.

• 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

• 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

• SIPv2: Session Initiation Protocol v2 (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

• G.711 (A-law and µ-law)

• G.726 (16/24/32/40 kbps)

• G.729 A

• G.723.1 (6.3 kbps, 5.3 kbps)

• Dynamic Payload

• Adjustable Audio Frames per Packet

• Fax Tone Detection and Pass-Through (Using G.711)

• Fax Pass-Though - Using G.711

• DTMF: In-band & Out-of-band (RFC 2833) (SIP Info)

• Flexible Dial Plan Support with Interdigit Timers and IP Dialing

• Call Progress Tone Generation

• Jitter Buffer - Adaptive

• Frame Loss Concealment

• Full Duplex Audio

• Echo Cancellation (G.165/G.168)

• VAD - Voice Activity Detection with Silence Suppression

• Attenuation / Gain Adjustments

• Flash Hook Timer

• MWI - Message Waiting Indicator Tones

• VMWI - Visual Message Waiting Indicator via FSK

• Polarity Control

• Hook Flash Event Signaling

• Caller ID Generation (Name & Number) - Bellcore, DTMF, ETSI

• Music on Hold Client

• Streaming Audio Server - up to 10 sessions

• Password Protected System Reset to Factory Default

• Password Protected Admin and User Access Authority

• Provisioning/Configuration/Authentication:

• HTTPS with Factory Installed Client Certificate

• HTTP Digest - Encrypted Authentication via MD5 (RFC 1321)

• Up to 256-bit AES Encryption

• Web Browser Administration & Configuration via Integrated Web Server

• Telephone Key Pad Configuration with Interactive Voice Prompts

• Automated Provisioning & Upgrade via HTTP, TFTP

• Asynchronous Notification of Upgrade Availability via SIP NOTIFY

• Non-intrusive, In-Service Upgrades

• Report Generation & Event Logging

• Stats in BYE Message

• Syslog & Debug Server Records - Per Line Configurable

• Per Line and Purpose Configurable Syslog and Debug Options

• 2 100baseT RJ-45 Ethernet Port (IEEE 802.3) -- 1 WAN, 1 LAN

• 1 RJ-11 FXS Phone Ports - For Analog Circuit Telephone Device (Tip/Ring)

• 1 RJ-11 FXO Phone Ports - For a Telco or PBX Connection

• Ring Voltage: 40-55 VRMS Configurable

• Ring Frequency: 10 Hz - 40 Hz

• Ring Waveform: Trapezoidal and Sinusoidal

• Maximum Ringer Load: 3 REN

• On-hook/off-hook Characteristics:

• On-hook voltage (tip/ring): -50 V NOMINAL

• Off-hook current: 25 mA min

• Terminating Impedance: 8 Configurable Settings including

• North America 600 ohms, European CTR21

• FCC (Part 15 Class B), CE, ICES-003, A-Tick Certification, RoHS

• DC Input Voltage: +5 VDC at 2.0 A Max.

• Power Consumption: 5 Watts

• Switching Type (100-240v) Automatic

• Power Adapter: 100-240v - 50-60Hz (26-34VA) AC Input

• Power, Internet, Phone 1, Phone 2

• Quick Installation, User, and Configuration Guides

• are downloaded from www.Linksys.com

• Administration Guide - Service Providers Only

• Provisioning Guide - Service Providers Only

• 3.98 x 3.98 x 1.10 in. (101 x 101 x 28 mm)

• 5.11 oz. (0.145 kg)

• 32ºF to 113º F (0ºC to 45º C)

• -13ºF to 185º F (-25ºC to 85º C)

• 10 to 90% Non-condensing, operating and non-operating

BlueWolf’s Alfa team are fully trained to install VoIP PBX systems. We cater for large commercial sites and small business offices.

Contact us for free advice and an obligation free quote.