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FXO FXS Analog Gateway – UCM200-DS Series
XNTEL UCM200-D2S2O2-Port FXS + 2-Port FXO Analog VoIP Gateway
Connect Analog Telephony with Modern IP Networks
The XNTEL UCM200-D2S2O is an FXO/FXS Analog VoIP Gateway designed to connect analog phones, fax machines, PBX systems, and PSTN lines with modern IP-based communication networks. With 2 FXS and 2 FXO ports, it provides a flexible bridge between traditional telephony and SIP-based VoIP environments.
Supporting IP-to-PSTN and PSTN-to-IP call routing, the UCM200-D2S2O helps businesses integrate existing analog infrastructure with VoIP without replacing their complete telephony setup. With features such as SIP support, T.38 fax, Caller ID, advanced call routing, and web-based management, it delivers reliable and efficient communication for modern business environments.
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Easy to Install
Clear
Recording
Online
Monitoring
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.
- Link VoIP networks to analog phone lines or PABX extension interfaces.
- High-quality voice calls with exceptional dependability and performance.
- This creative approach is scalable to businesses of any size.
- Caller ID presentation and restriction
- Hotline extension setting
- Web-based remote administration
- Restrict unwanted calls with a list of prohibited numbers
- VoIP connectivity with high performance for SMBs
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XNTEL's 2 ports analog FXS/FXO gateway provides a cost-effective and straightforward means of assisting traditional telephone, fax, and PBX systems to link with IP networks.
Salient Features
Cost-Effective Solution
XNTEL's 2-port FXS/FXO gateway offers an affordable and simple method for connecting legacy phone, fax, and PBX setups to modern IP networks.
Versatile Deployments
It empowers multi-branch organizations and call centers to implement powerful, flexible VoIP solutions while significantly reducing expenses.
Seamless Protocol Conversion
When integrated with a WAN, LAN, or PBX, the device converts standard analog PSTN signals into IP-compatible formats for easy network transmission.
Unified Messaging Support
The UCM200-D2S2O 4-Port Analog VoIP Gateway is specifically designed to handle unified messaging and voicemail applications effortlessly.
Vintage PBX Compatibility
Users can easily link their older PBX hardware directly to a local area network (with optional 10/100/1000M speeds) using a standard Base-T Ethernet port.
Advanced Analog Signaling
Its analog loop start capabilities fully support in-band signaling (such as DTMF or FSK), serial protocols, and T.38 for Fax over IP functionality.
FXS FXO VoIP Gateway Features
| Fully scalable framework with a non-blocking architecture | Seamlessly integrates into current telephony hardware |
| Features open-standard SIP compatibility with multi-proxy server registration | Facilitates inbound and outbound IP calling via standard analog lines |
| Supports call budgeting by total duration, allocated funds, and call volume | Provides configurable routing between TDM-to-IP and IP-to-TDM networks |
| Blocks unapproved inbound connections using a denied numbers list | Transmits real-time call logs directly to a CDR server |
| Manages the display and blocking of caller identification | Allows configuration for dedicated hotline extensions |
| Enables remote system management via a web interface | Grants console administration access through SSH and Telnet protocols |
| Technical Specification : | ||
|---|---|---|
| Physical Interface | UCM200 H Series | UCM200-D2S2O |
| Phone Interface: 4 Ports FXS/FXO, RJ-11(some models are RJ21 available) | ||
| Ethernet Interface: 2* RJ-45 10/100Mbps Base-T Ethernet, Female RJ-45 | ||
| 1000M LAN/WAN available for some product models while required | ||
| Session Capacity | 4 SIP channels (UCM200-D2S2O) | |
| 2 FXS/ FXO channels (UCM200-D2S2O) | ||
| Connectivity | Supported Dial Mode: DTMF and Pulse | |
| Pulse: 10 and 20PPS | ||
| Caller ID:DTMF/FSK | ||
| Max Cable Length:5KM | ||
| Reversed Polarity | ||
| Open VPN | ||
| VoIP Protocols | TLS / SRTP ;OpenVpn;SIP V2.0 (RFC 3261,3262,3264);IMS/3GPP; SDP ;REFER (RFC 3515);RTP/RTCP;STUN (RFC3489) ;ARP/RARP (RFC 826/903);SNTP (RFC 2030);DHCP/PPPoE;TFTP/HTTP/HTTPS;DNS/DNSSRV (RFC 1706/RFC2782);VLAN802. 1P/802.1Q | |
| Call & Routing | T.38/Pass-through, up to 14.4kbpsPort Groups | |
| IP Trunks | ||
| Primary and Secondary SIP Account | ||
| 4 Inbound/Outbound Routing | ||
| Number Manipulation | ||
| Digit maps | ||
| TDM to IP or IP to TDM | ||
| IP load balancing | ||
| IP fault tolerance | ||
| Voice Capability | G.711A/U law, G.723.1, G.729A/B,G.726,iLBC,AMRG.711A/U law, G.723.1, G.729A/B,G.726,iLBC,AMR | |
| Comfort Noise Generation(CNG) | ||
| Echo Cancellation(G.168) | ||
| DTMF mode: Signal/RFC2833/INBAND | ||
| Silence suppression with comfort noise | ||
| G.168 automatic echo cancellation | ||
| Call Progress Analysis (CPA), including Positive Voice Detection, Positive Answering | ||
| Machine Detection (PAMD), DTMF detection, and fax tone detection | ||
| Manageable based call routing TDP-IP/IP-TDM. | ||
| Restrict unwanted calls with list of denied numbers. | ||
| Voice Activity Detection (VAD) | ||
| Adaptive (Dynamic) Jitter Buffer | ||
| Programmable Gain Control | ||
| Hook Flash | ||
| Network Capability | Static IP, PPPoE, DHCP Client | |
| IPv4, IPv6 | ||
| Static/dynamic ARP | ||
| DIFFServ, ToS | ||
| NAT (Rout and Bridge)+ | ||
| MAC Address Clone | ||
| Static routing+ | ||
| Built-in Firewalls | ||
| QoS, Traffic Shaping | ||
| Voice/Data/Management Vlan | ||
| Maintenance & Upgrading | SNMP/TR069. | |
| Auto Provision | ||
| Action URL | ||
| Digit map | ||
| Web/Telnet. ACL | ||
| Configuration Backup/Restore | ||
| Bandwidth Optimization | ||
| Routing Rules based Prefixes | ||
| Firmware Upgrade via WEB | ||
| Syslog and CDR. | ||
| Access Rule list. | ||
| Network Capture | ||
| Outward Test(GR909). | ||
| Automatic Time Synchronization | ||
| IVR local Maintenance. | ||
| Cloud-based Management | ||
| Caller/Called Number Manipulation | ||
| Open-standard SIP support and register to multiple SIP proxy servers. | ||
| Application Capabilities | SupportedDial Mode: DTMF and PulseCall waiting | |
| Blind Transfer | ||
| Attend Transfer | ||
| Call forward on Busy | ||
| Call forward on No Reply | ||
| Unconditional Call Forward | ||
| HotlineCall hold | ||
| DND | ||
| Call Pickup | ||
| 3-way conference | ||
| Voicemail | ||
| Conferencing Resource | Call budgeting based on allocated amount, minutes and call count | |
| Complete non-blocking architecture and Scalable System | ||
| Hotline extension setting | ||
| Support 3-Way and Multi-Way Conferencing | ||
| Environment & Power | Power Supply: 100-240V, 50-60Hz+ | |
| Power Consumption: Approximately 50W | ||
| (Storage): -20 ~85°C | ||
| Humidity: 10%-90% No condensation. | ||
| Operating temperature range: -10 °C ~ 55 °C | ||
| Physical Dimension | L*W*H 140(mm)*100(mm)*30(mm) | |
| Weight Approximately 0.44ibs(about 0.2kg) | ||
| Warranty/Certifications | 3 years: The first year exchange for free. On the Second & Third free to repair. CE, FCC or Any other Certificates Customizable Broadsoft, Elastix, Asterisk, Teams and other UC platform | |
ONE SOLUTION FOR EVERY INDUSTRY.
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