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VoIP-FXS ATA

Introduction

A VoIP-FXS ATA offers integrated VoIP and FXS Port. Users can plug one or more standard analog telephone devices into the ATA’s FXS ports and the analog device(s) will connect transparently to the IP network.  An ATA provides the user with the ease of using a standard telephone instrument, yet make VoIP calls. Since the ATA communicates directly with the VoIP server, it does not require a personal computer or any software such as a softphone.
 

Application Diagrams

fx-ata
The ATA offers FXS interfaces for connecting the analog telephone instrument or PBX CO port. The ethernet port on ATA connects to the IP network. Now using the analog terminal, a user can establish IP calls. When connected to the CO Port of a legacy PBX system, the PBX users may share the new VoIP line for making IP calls. An ATA effortlessly bridges one to the IP network. Using an appropriate VoIP service provider, long distance or inter-office call charges can be reduced significantly or eliminated through peer-to-peer calling on the IP network.

Making an outgoing call is as easy as from a normal telephone. Call progress tones like Dial Tone, Ring Back Tone and Busy Tone are fed to the caller as per the called number status. The FXS ports can make outgoing calls on a common or different SIP accounts. In addition, number based SIP account selection is provided to select the most economical SIP account for a given outgoing number. An incoming call from a SIP account can be routed to any one or both FXS ports. All different CLIP protocols are supported so that the user can identify the caller before answering the call. Once a call is established, features like Call Hold, Call Toggle, Call Transfer, Call Wait and Conference are supported to manage two calls from the same FXS port. Call forward in different conditions and Do Not Disturb is also provided.

These ATAs also cart two Ethernet ports - one for WAN and the other for LAN. The user can connect his PC on the LAN port and browse the internet or check his emails while talking on VoIP calls
 

Features List

Software Features

100Rel/PRACK (RFC 3262)
Answer Signaling
Auto Configuration Auto PSTN Fallback
Automatic Number Translation Called Party Number Table
CLIP (DTMF, FSK-ITU-T V.23, Bellcore 202A) Comfort Noise Generation
DHCP Client/ PPPoE Dialed Number Table
Digest Authentication Disconnect Signaling
Echo Cancellation Fax over IP-T.38 and Pass Through
Flash Timer Forward Error Correction (FEC)
Full Duplex Audio Incoming Call Routing
LED Indications MAC Cloning
Multiple Gateway Support Multistage Dialing
Password Protection PCAP Trace
Peer-to-Peer Calling PIN Authentication
Polarity Reversal PPPoE
Programmable Call Progress Tones and Rings SIP over TCP
Speech Volume Setting (Transmit and Receive) Speed Dialing
STUN Symmetric RTP
Syslog Client Voice Activity Detection
VLAN Tagging Web based GUI for Configuration
Supplementary Services
Call Forward Unconditionally Call Forward on Busy
Call Forward on No Reply Call Hold
Call Toggle Call Waiting
Caller ID Call Transfer-Blind
Call Transfer-Attended Conference 3 Party
Do Not Disturb (DND) Making Second Call