VoIP-FXO-FXS ATA
Introduction
A VoIP-FXO-FXS ATA offers integrated VoIP, FXO and FXS Ports. 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. Further, the FXO port provides connectivity to a CO line. Connecting a CO line to the FXO port of ATA and the Ethernet port to the VoIP network, a user can place both IP and fixed-line calls from a single telephone instrument.
Application Diagrams
The FXS port is used for connecting an analog telephone instrument or the CO port of existing PBX system. 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. The ATA also offers features like Automatic Number Translation, which efficiently transforms a number to a string understandable by the destination network. In case of no IP connectivity, the POTS line can acts as the failover fallback line.
The FXO port allows users to originate a call and terminate it at VoIP network. The routing logic used between FXO and VoIP helps the user to dial a destination PSTN number directly over VoIP network. Using FXO VoIP routing logic two or more PBX can be networked to offer seamless connectivity between the PBXs..
The FXO port allows users to originate a call and terminate it at VoIP network. The routing logic used between FXO and VoIP helps the user to dial a destination PSTN number directly over VoIP network. Using FXO VoIP routing logic two or more PBX can be networked to offer seamless connectivity between the PBXs..
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 |
