ATA171 plus/ATA172 plus

SIP IP ATA for Analog Phone device
  • Support up to 5 Register Accounts at five SIP Servers
  • Compatible with VGCP Encryption protocol to avoid from blocking
  • Suit to end user use with Voice over IP Service Provider
  • Auto Provision feature
  • Support PPTP VPN Client
  • WAN and LAN support NAT feature

Introduction
Information
DataSheet
Documents
ATA171 plus (high res)
ATA172 plus (high res)

ATA172plus and ATA171plus are two analog phones and one analog phone set SIP device which allow user to make or receive VoIP call through Internet Telephony service provider. This device is suitable for single user for ITSP service provider to install at home or office with affordable price and convenience installation.

To select freely up to 5 SIP service Accounts

ATA is appropriate to use for VoIP Service Providers, IP Centrex service and IP-PBX within offices and remote branch offices. Up to 5 SIP Servers ( or ITSP Service provider or alternative IP-PBX ) can be configured at both ATA171plus and ATA172plus simultaneously. You can dial one of five accounts number directly no hassle.

Provision is easier than before

Auto Provision server software installs at Windows or Linux platform are supported to manage, configure and configure firmware download remotely to ATA. It is a convenience device for VoIP ITSP Service provider to manage ATA easily.

Suit to IP Telephony Service Provider

Both ATA171plus and ATA172plus are SIP IP device to connect with existing analog telephone set to make IP call. Its compact design and easy installation allow home user or single user to make or receive call just like an legend telephone call but less cost. It is compatible with broadband internet service device such as ADSL/Cable Modem and WiMax/3G Modem. It help ISP provider to provide Telephony service to existing customer without additional cable. It provides fast, easily, cost effective and remotely management feature to migrate from ISP to Telephony ISP service.

Specification
  • Interface:
    • Ethernet port (RJ-45, 10/100 base-T)
      • 1-WAN port, connect to IP Network
      • 1-LAN port connect to PC with NAT
    • DC +12V power input Jack
    • Reset key to return Factory setting
  • IP Network connection
    • IPv4 (RFC 791),
    • MAC Address (IEEE 802.3)
    • MAC Clone Setting
    • IP/ICMP/ARP/RARP/SNTP
    • Static IP
    • DHCP Client (RFC 2131), WAN port
    • DHCP Server, LAN port
    • Wire line speed more than 85MB at Bridge mode
    • PPPoE
    • DDNS
    • DMZ
    • VLAN : 802.1Q/1P
    • Virtual Server (DHCP Server IP range)
    • DNS Client
    • PPTP VPN Client tunnel 64 bits without compression
    • SNTP support Daylight Saving Time (DST) configuration
    • SNTP with time zone
    • TCP/UDP (RFC 793/768)
    • RTP/RTCP (RFC 1889/1890)
    • IPV4 ICMP (RFC 792),
    • TFTP Client
    • QoS Support : ToS
  • SIP Protocol :
    • RFC3261 compliance
    • Support up-to 5 SIP Register Accounts
    • SIP Proxy compatible with brand name : Asterisk and Nortel
    • SIP UDP Protocol
    • Support SIP compact Form
    • SIP Session Timer (RFC 4028)
    • MD5 Digest Authentication (RFC2069/RFC2617)
    • Message Waiting Indication (RFC3842)
    • Event Notification (RFC3265)
    • REFER (RFC3515)
    • Support Outbound Proxy
    • Support DNS SRV to locate SIP Server (RFC 3263)
    • Support STUN NAT Traversal
    • Support “rport” parameter (RFC 3581)
  • Audio Codec :
    • G.711 A-law/μ-law, G.729, G.723, G.726
    • Silence Suppression
    • VAD/CNG
    • Jitter Buffer : Up to 32 packets
    • LEC : Line Echo Canceller
    • Packet Loss Compensation
    • Automatic Gain Control
    • In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
    • Adaptive/Configurable Jitter Buffer
    • Acoustic Echo Cancellation
    • Speed Dial
    • Phone Book ( up to 140 records )
    • Clock, Call-Duration display
    • Call History of Missed, Received and Dialed
    • Dialing Plan with drop, replace, Insert dialing digits
    • Selectable Call Progress Tone
    • Support Personal Melody Ring
    • Auto Answer Mode
    • Support Specified Line Calling
  • Call Features :
    • Caller ID display DTMF (before/after 1st ring) and FSK (before 1st ring )
    • Tone Generation: Ring, Ring Back, Dial, Busy, call waiting and 
      congestion tone
    • Out-Band DTMF : RFC2833 and SIP Info
    • Voice Mail with Indication
    • Speed Dialing
    • Call Waiting/Switching between Calls
    • Call Forward (Busy, Unconditional, No Answer)
    • DND : Always ON or configurable period
    • Call Hold
    • Call Mute
    • Call Transfer
    • Flexible Dial Plan: Drop and Replace Rule
    • T.38 FAX : Enable or G.711 Codec A-law/u-Law Pass trough Codec
    • Alarm Ring Reminder
    • 3-way conference call
    • Music-on-hold support (via IPPBX or local)
    • Redial
    • Hot Line
    • Support Peer to Peer Dialing
    • Volume Adjustment: Handset Volume (receiver) and Handset Gain 
      (Transmitter) selection
    • Flash Time Detection: range from 70 to 2550 ms
    • ON-HOOK Voltage -48Vdc
    • Support 12/16Khz metering signal or Polarity reversal for Billing
    • Service Up to 1 Kilo-meter distance from ATA to analog telephone set
    • Global Country Impedance setting
    • CPC Delay : 2 to 5 seconds (Open Loop Disconnect time)
    • CPC duration: 10 to 1200ms 
  • MANAGEMENT :
    • Administrative Telnet CLI and HTTP
    • 2 Levels of User Access Right with Password protection
    • Management from WAN enable or disable
    • Provides System Status Logs
    • Network Status Display : WAN and LAN port Status
    • Diagnostics (debug through syslog)
    • Configuration Backup/Restore
    • Firmware configurable updated
    • Reset to factory Default
    • Support Auto Provision through MAC address
    • Voice configuration from analog telephone set with DTMF tone and voice 
      announcement

    • ** Support Welltech and VGCP proprietary encryption protocol at SIP 
    Signal and Voice codec during transmitting to IP network in order to 
    Anti-ISP block of VoIP call. This feature only be available with Welltech 
    SIP server or SIPPBX6200 IP-PBX

  • Environmental :
    • Dimension: 9.9(H) × 9.9(W) × 3.2(T) CM
    • Weight: 0.35kg (One unit with packing)
    • Operating Temp. & Humidity
      • Temp.: 0°C~45°C (32°F~113°F)
      • Humidity: 10%~90% relative humidity, non-condensing
    • Power Adaptor:
      • INPUT: AC100V~240V, 50/60Hz
      • OUTPUT: DC 12V,
  • Approvals:
    • CE, FCC, LVD and RoHS
  • Country of origin:
    • Made in Taiwan
  • Packing Accessories
    • ATA171plus or ATA172plus x 1 pcs
    • AC to DC+12V Power adaptor x 1 pcs
    • 1 meter Ethernet cable x 1 pcs
    • CD User Manual x 1 pcs
  • Warranty
    • One year

Ordering Information
  ATA171 plus ATA172 plus
Analog Phone 1 line FXS 2 lines FXS
Color Gray Milk White