
FXO-FXS VoIP Gateway - UCM200-D8S8O
8-Port FXS + 8-Port FXO Analog VoIP Gateway
Bridge Legacy Telephony with Modern VoIP Networks
The XNTEL UCM200-D8S8O is a high-capacity FXS/FXO Analog VoIP Gateway designed to connect traditional analog telephony equipment, legacy PBX systems, fax machines, and PSTN lines with modern IP-based communication networks.
With 16 telephony ports and up to 16 sessions, the UCM200-D8S8O provides a scalable communication solution for businesses that need to integrate existing analog infrastructure with VoIP. It supports SIP/TDM signaling and IP ↔ PSTN call routing, enabling communication between traditional switched networks and IP-based destinations.
The gateway enables organizations to modernize their telephony infrastructure gradually, protecting their investment in existing equipment while introducing VoIP connectivity and IP-based communication applications. It is suitable for SMBs, call centers, branch offices, multi-location businesses, and enterprise telephony environments.
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FXO-FXS VOIP GATEWAY - UCM200-D8S8O
FXO VoIP Gateway Benefits :

Feature Packed
A complete features set that meets all of your requirements and lowers communication costs.
User-Friendly
Easy to use Web GUI and intuitive control. Effortless intallations, configure, debug and maintain.
Elevated Efficiency
Excellent voice compression using established codecs for the industry. Compatible with various IP systems.
Adaptive Setting
XNTEL PBX supports standard open protocol TR-069 and PPPoE or DHCP provisioning.
What is FXS and FXO Ports ?
FXS and FXO are the ports of electronic device which is used by POTS, often as an analog phone line or old telephone service. The two ports are constantly connected together in order to placing a call.
- FXS (Foreign Exchange Subscriber) port is used to transmit the internal connection to office device such analogue fax machines and analog phones. To provide dial tone, ringing voltage, and battery power to FXO devices through a wall jack or FXS interface.
- The FXO stands for Foreign exchange office that is connected to an external phone line via FXO port. On the desk phone, analog fax machine, or modem, there is a plug called an FXO port. As a result, the FXO interface receives the POTS service from the PSTN Central Office and creates a loop closure.

The FXS and FXO ports must be linked in order to make a call. Both incoming and outgoing calls are affected by this. You dial a number and use the PSTN service to establish a connection to the other end when you pick up an analog phone (an FXO device) that is linked to an FXS port. When you use your office phone to call a number, the DTMF (Dual-Tone Multi-Frequency) digits are encoded and sent to an FXS port.
The connected FXO device will begin ringing when an incoming call is received over the FXS interface, which will also send the ringing voltage. Line establishment occurs once the phone is picked up. On the other hand, communication will not work when two FXO or two FXS ports are connected. No dial tone will be delivered, even if you desire to connect two internal FXO devices.
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FXO-FXS VoIP Gateway - UCM200-D8S8O

The Aegis XNTEL FXS-FXO VoIP Gateway delivers a dependable, budget-friendly solution to seamlessly bridge your existing analog telephones, fax machines, and traditional PBX hardware with modern IP networks. By integrating with the CM300G series—which offers a powerful array of gateway options supporting 1 to 4 FXS ports and 4 SIP channels—organizations can maximize their current telephony investments while executing a smooth, step-by-step transition to VoIP. The CM300G analog gateway connects directly to a PBX or LAN, converting standard analog PSTN signals into optimized formats for IP transmission, providing call centers and multi-location businesses with a highly capable, flexible, and cost-efficient communication upgrade.
Salient Features
Fit to Diverse Applications
Used to integrate network-hosted apps with the PBX, expand the PBX to branch offices, link IP phones to a legacy PBX, and integrate different voice and call processing capabilities in an enterprise LAN or WAN environment.
Flexible Call Routing
Calls can be routed from the switched network to a VoIP destination on the IP network, or vice versa. These call routing choices include TDM to IP or IP to TDM, including IP load balancing and IP fault tolerance.
Controlled Migration to IP
Compatible with a wide range of popular PBX manufacturers, generic FXO/FXS lines, and digital PSTN lines (compliance would be provided), protects investment in legacy telephony equipment.
User-Friendly Management
Supports configuration via serial, Telnet, and a web browser including context-sensitive help Easy to install, configure, debug, and maintain.
Enhanced Voice/Fax Processing
Supports a variety of compression algorithms, including G.711 A-law and μ-law, G.729A/B; Transcode fax from T.30 fax protocol (supporting V.17) to T.38 for transmission over a packet network.
Web Server Interface & Hot Swap
Every gateway unit comes with a web server interface that enables setup and software updates using a web browser; this feature enables the addition or removal of gateway units without affecting other gateways.
FXS FXO VoIP Gateway Features
| Completely non-blocking architecture and Scalable System | VoIP connectivity with high performance for SMBs |
| Easy integration with existing telephony interfaces | Voice optimization for enhanced user experiences |
| Place and receive IP calls from analogue extensions | Easy to install, configure and maintain |
| Support for the open-standard SIP protocol and registration on several SIP proxy servers | enhanced call routing capabilities with excellent voice quality |
| Restrict unwanted calls with a list of prohibited numbers | Build-in firewall and access rules |
| Caller ID presentation and restriction | Bandwidth optimization and cloud-based management |
| Hotline extension setting | Support SIP and other customizable protocols |
| Web-based remote administration | Backup SIP Servers |
| Technical Specification : | ||||
|---|---|---|---|---|
| Main Model No. | UCM200 H | UCM200 H | UCM200 H | UCM200 H |
| CM300G-D2S2O | CM300G-D4S4O | CM300G-D8S8O | 300G-D16S16O | |
| Sessions | 4 | 8 | 16 | 32 |
| Telephony Ports | 4 | 8 | 16 | 32 |
| Interface | RJ11 | RJ11 | RJ11, RJ21, RJ45 | RJ11, RJ21, RJ45 |
| LAN | 1*10/100Mbps | 2*10/100Mbps | ||
| Console Port | N/A | 1 | ||
| Signaling | SIP/TDM | |||
| Power Supply | Single | |||
| Power Requirements | 12V DC | AC Frequency 47 Hz to 63 Hz | ||
| Codecs | G.711/729/723/722/iLBC/AMR | |||
| DTMF Mode | RFC2833, SIP Info, In-band | |||
| Fax | T.38 and Pass-through | |||
| QoS | DiffServ, ToS, 802.1 P/Q VLAN tagging | |||
| Network | DHCP, PPPoE, Static Route, VLAN | |||
| Network Protocol | TCP/UDP, HTTP, ARP/RARP, DNS, NTP, TFTP, TELNET, STUN | |||
| Management Protocol | TR-069* | |||
| Signaling | FXS Loop Start, FXS Kewl Start | |||
| Caller ID | BELL202, ETSI (V23), NTT (V23-Japan), and DTMF-based CID | |||
| FXO Connectivity | Programmable AC Impedance, Hangup Detection, Answer Detection, Caller ID Detection | |||
| Dimensions H*W*D(mm) | 220*148*40 | 440*267*44 | ||
| Routing | Call routing(IP ↔ PSTN) | |||
| Mounting | Desktop | Wall-mount | ||
| Compatibility | Interoperable with most IP-PBX and UC Platform, and field-proven by SMB and Carriers Worldwide | |||
| Environment | Operating temperature range : 0 to +55 °C, 8-90% relative humidity non-condensing Storage temperature range : 20 to +85 °C, 8-90% relative humidity non-condensing | |||
| Safety | Compliant with most international standards, please ask sales representatives worldwide. We would comply all new safety standard to for different regions around the world while needed. | |||
| EMC/EMI | Compliant with most international standards. For compliance documents, please contact sales representatives. | |||
| OAM&P | Network Time Protocol(NTP) Web User interface (WebUI) supports configuration via browser SNMP MIBs | |||
ONE SOLUTION FOR EVERY INDUSTRY.


































