ATA 2 FXS ( Analog Telephone Adapter )

$60.00

Tax included

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Description
Product Details

Analog Telephone Adapter ATA172

Isi 2 Port Analog Phone

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