LinkSYS SPA 3000 Voip Adaptör

LinkSYS SPA 3000 Voip Adaptör
Ürün Kodu: LinkSYS SPA 3000 Voip Adaptör
Stok Durumu: Sorunuz
Fiyatı: 47.20$
KDV Hariç: 40.00$
Miktar:     ya da   A.Listeme Ekle
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Linksys SPA3000, PAP2 nin bir üst modelidir.

Voip hattınızla birlikte sabit hattınızı kullanmanız için bir tane sabit hat girişi bulunmaktadır.

Kurulumu ve kullanımı PAP2 ile neredeyse aynıdır.

Micro-Density ?PSTN Gateway? + ATA

The SPA-3000 continues to deliver on Sipura Technologys mission to provide market leading,

best-in-class VoIP end points providing freedom and opportunity to service providers and end users.

The SPA-3000 features VoIP adapter functionality found in the SPA-2000 and SPA-1000/1

with the additional benefit of an integral connection for legacy telephone network hop-on, hop-off applications.

SPA-3000 users will be able to leverage their broadband phone service connections more than ever by

automatically routing local calls from cell phones and land lines to a VoIP service provider and vice versa.

A typical user calling from a land line or mobile phone will be able to reduce and even eliminate

international and long distance telephone charges by first calling their SPA-3000 via a local

phone number or by using a telephone connected directly to the unit.

The advanced authentication and call routing intel-ligence programmed into the SPA-3000

will connect the caller via the Internet to the far end destination with security and ease.

Using the SPA-3000 at the far end, calls can be answered immediately or further processed as a local

call to any legacy land line or mobile phone allowed by the SPA-3000 dial plan.

If power and/or IP network connectivity is lost to the unit or the VoIP service is down

calls can be sent to a traditional carrier via the FXO interface.

Linksys SPA3000 Voip Adapter FXS FXO PSTN

VoIP to PSTN (USA) Service Call Origination and Termination

PSTN (USA) to VoIP Service Call Origination and Termination

Single or Dual Stage Dialing

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

Per Call Authentication and Associated Routing

Least Cost Routing Support

Terminating Impedance Agnostic – 8 Configurable Settings

Call Waiting, Cancel Call Waiting

Caller ID Detection (Bellcore Type 1)

Caller ID with Name/Number (Multinational Varients) Display

Caller ID Blocking

Call Waiting Caller ID with Name/Number

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 and Call Back on Busy

Call Blocking with Toll Restriction

Distinctive Ringing

Off-Hook Warning Tone

Selective/Anonymous Call Rejection

Hot Line and Warm Line Calling

Speed Dialing of 8 Numbers/Addresses

Music on Hold

Forward Calls to VoIP service – Selective, Authenticated, All

Forward Calls to PSTN service – Selective, Authenticated, All

PSTN Line Sharing with Multiple Extensions

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

Advanced Inbound and Outbound Call Routing

Single Stage and Two Stage Dialing

Independent Configurable Dial Plans – Up to 8

Force PSTN Disconnection

Sequential Dialing Support

VoIP to PSTN Authentication and Routing

VoIP to PSTN Gateway Enable/Disable

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

VoIP PIN Max Retry Setting

One Stage Dialing 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
  • Caller Auth Method: None, PIN, Caller ID
  • Ring Through to FXS Enable/Disable
  • Ring Through Tone – Configurable
  • Caller ID (Bellcore 1) for VoIP Service Access
  • 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 (FXO Behavior)
  • 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
  • CPC (Removal of Tip/Ring Voltage Momentarily)
  • Polarity Reversal
  • Long Silence (Confugrable Time Setting)
  • Disconnect Tone (e.g. Reorder Tone)
  • 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

    ( Regulatory Compliance)
  • FCC (Part 15 Class B and Part 68) , CE , ICES-003

  • Password Protected System Reset to Factory Default
  • Password Protected Admin and User Access Authority
  • Provisioning/Configuration/Authentication:
  • HTTP Digest – Encrypted Authentication via MD5 (RFC 1321)
  • Up to 256-bit AES Encryption

  • Quick-Start Installation and Configuration Guide
  • User Guide
  • Administration Guide – Service Providers Only
  • Provisioning Guide – Service Providers Only

    (Package Contents)
  • 1 – SPA3000 Phone Adapter Unit
  • 1 – 5v Power Adapter
  • 1 – RJ45 Ethernet Cable
  • 1 – RJ11 Telephone Cable

(Dimensions:) 4.05 x 1.1. x 3.75 in (102.87 x 27.94 x 95.25 mm) W x H x D
(Unit Weight): 2.15 lb (0.975 kg)
(Operating Temp).: 32~113F (0~45C)
(Storage Temp): -13F~185F (-25~85C)
(Operating Humidity): 10~90% Non-condensing
(Storage Humidity): 10~90% Non-condensing

(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)
  • 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)
  • SNTP – Simple Network Time Protocol (RFC 2030)

(Voice Gateway)

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

    Voice (Algorithms )
  • 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 (Fax Capability)
  • Fax Tone Detection and Pass-Through (Using .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

    Provisioning, Administration & Maintenance:

  • Web Browser Administration & Configuration via Integrated Web Server
  • Telephone Key Pad Configuration with Interactive Voice Prompts
  • Automated Provisioning & Upgrade via HTTPS, 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

(Physical Interfaces):

  • 1 10baseT RJ-45 Ethernet Port (IEEE 802.3)
  • 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

FXS: Subscriber Line Interface Circuit (SLIC):

  • Ring Voltage: 40-55 VRMS Configurable l*
  • Ring Frequency: 10 Hz – 40 Hz l*
  • Ring Waveform: Trapezoidal and Sinusoida l*
  • 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

Power Supply:

  • 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, 1.8m cord

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