ASTERISK IP-PBX


VOIP PHONES

GATEWAYS

 
 

 

ATA (VoIP)


ATA-171/172/171P

ATA - The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it is a very straightforward setup.

The ATA-17x series contain four models of gateway products: ATA-171, ATA-172, ATA-171P. With outstanding design and dazzling appearance, ATA-17x series can satisfy all users and meet their different requirements. ATA-171/172 is one/two-port analog telephone adapter, and user can connect with one/two analog phone set to enjoy VoIP application. ATA-171P is one-port analog telephone adapter plus one PSTN backup lifeline, which allows user to dial and receive PSTN/VoIP call in one identical phone set.

  Order Information  
Model
Interface Specification
Color
171
One Port FXS
172
two Port FXS
172
One Port FXS + one PSTN backup Line port

Physical interface

  • RJ-45
    A. LAN X 1 for connecting to HUB or
    ATU-R directly
    B. PC X 1 for PC connection
  • RJ-11
    A. Phone X 1 for ATA-171
    B. Phone X 2 for ATA-172
    C. Phone X 1, Line X1 for ATA-171P
  • Dimension: 9.9 X 9.9 X 3.2 cm

Network and Protocol

  • SIP v1(RFC2543), v2(RFC 3261)
    A. Outbound proxy
    B. Support backup proxy registration
    C. Support IP or domain name for
    primary and secondary proxy address
    and auto switching is enabled.
  • IP/ICMP/ARP/RARP/SNTP
  • TFT Client/DHCP Client/PPPoE Client
  • Telnet/HTTP Server
  • DNS Client
  • NAT transversal
    A. STUN
    B. UPnP
  • Support ToS
  • Security
  • A. HTTP 1.1 basic/digest authentication
    for Web setup
  • B. MD5 for SIP authentication
    (RFC2069/RFC2617)

    Call function

  • Call Hold
  • Call Waiting
  • Call Forward
  • Caller ID
  • Flash
  • Volume Adjustment
  • Speed dial key
  • Phone book

Voice feature

  • Voice codec
    A. G.711: 64k bit/s (PCM)
    B. G.726: 16k/24k/32l/40k bit/s
    (ADPCM)
    C. G.729A: 8k bit/s (CS-ACELP)
    D. G.729B: adds VAD & CNG to
  • G.729
  • VAD : Voice activity detection
  • CNG : Comfortable noise generation
  • LEC : Line echo canceller
  • Packet Loss Compensation
  • DTMF
    A. In-Band DTMF
    B. Out-Band DTMF
    C. SIP Info
  • Tone generation
    A. Ring Tone
    B.Ring Back Tone
    C.Dial Tone
    D.Busy Tone
    E.Programming Tone

Firmware and configuration update

  • Web Browser
  • Telnet
  • Voice configuration
  • TFTP
  • HTTP

Environment

  • AC Power
    A. 110~220V ħ 10V
    B. 60Hz ħ 3Hz
  • Environment
    A. Temprature:0oC~40oC
    B. Hmidity:10%~90%RH
 
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