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SIP to PRI Converter System — XNTEL UCM300D-2E1

High-Capacity SIP-to-PRI Trunk Gateway

The XNTEL UCM300D-2E1 is a reliable and robust SIP-to-PRI Converter System designed to connect modern IP communication networks with existing ISDN PRI/E1/T1 and legacy PBX infrastructure. It enables businesses to route calls between traditional fixed-line telephony and cost-effective IP networks, supporting a smooth transition to VoIP without requiring major changes to existing communication infrastructure.

With support for 2 E1 interfaces and up to 60 simultaneous VoIP-to-ISDN calls, the UCM300D-2E1 is suitable for corporate enterprises, SMEs, call centers, telecom operators, and system integrators that require scalable and reliable SIP-to-PRI connectivity.

The gateway uses dedicated DSP chipsets for IP/TDM signal processing and voice-quality optimization, while its compact desktop design, software-configurable E1/T1 ports, and cost-effective call-routing capabilities make it practical for a wide range of telephony deployments.

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Easy to Install

Clear
Recording

Online
Monitoring

SIP TO PRI CONVERTER SYSTEM - UCM300D-2E1

Our SIP-to-PRI converter is reliable and robust, allowing businesses to seamlessly integrate new communication technologies with their current phone systems

SIP-to-PRI Trunk Gateway UCM300D-2E1 Benefits :

SIP TO PRI CONVERTER SYSTEM - UCM300D-2E1

Feature Packed

A complete features set that meets all of your requirements and lowers communication costs.

Easy to Install

You can save a tonne of time with a simple installation and an easy-to-use GUI.

Compatibility

Compatible with various ISDN PBX, IP-PBX, EPABX and softswitch.

Reliability

Stable with advanced hardware and software architecture.

  • Support 2 E1
  • Support SIP and SS7/ISDN/R2/CAS and More
  • Up to 60 simultaneous VoIP to ISDN calls
  • Superior Voice Quality and Reliability in Full Load
  • Compact in size
  • Cost-effective call routing
  • Software configurable E1/T1 Ports
  • Reliable robust market proven
  • Rugged design for 24x7 scenario

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Digital Gateway PSTN and Internet Network Architecture

The XNTEL SIP to PRI Converter System (UCM300D-2E1) Trunk Gateway enables seamless call routing between traditional ISDN PRI/E1/T1 lines and modern, affordable IP networks, allowing organizations to capitalize on highly economical VoIP services. By integrating this gateway with your current legacy PBX setups, businesses of any scale can significantly cut voice communication expenses without requiring massive upfront investments to overhaul their existing infrastructure. The system sets itself apart from market alternatives through the inclusion of specialized DSP chipsets, which expertly handle IP/TDM signaling while maximizing audio clarity. Thanks to its exceptional dependability and accessible price point, the UCM300D-2E1 model serves as an ideal bridge solution for access networks across corporations, small to medium enterprises, call centers, and telecommunication providers.

Why Choose Aegis XNTEL E1/PRI Trunk Gateway

Reliable & Robust Telco-Grade Architecture

Built with an external power source, industrial-grade heat sinks, dual 100M Ethernet ports, and carrier-grade hardware, this gateway guarantees a 200,000-hour Mean Time Between Failures (MTBF) and flawless long-term operation.

Dedicated DSP-Empowered Capability

Features standard, built-in signaling packets (including ISDN, SIP, CAS, and R2) alongside highly adaptable protocols perfectly suited to interface with global carrier networks.

99.9999% Availability in Full Load

Each unit successfully supports up to 2E1 (60 channels) while maintaining exceptional responsiveness, system stability, and peak performance—even when operating under demanding loads or limited bandwidth constraints.

Highly Adjustable For Diverse Soft Switch

Powered by proprietary core technologies, it ensures flawless compatibility with a vast range of PBX systems and softswitch platforms, including Avaya, Mitel, Asterisk, Broadsoft, FreeSwitch, and many others.

SIP to PRI Trunk VoIP Gateway Features

Key Features:
BlocklistCall Duration Limitation
Caller ID PrefixCall Detail Record (CDR)
Caller IDConfigure backup/restore
Call Routing RulesCustom Prompts
E1/T1/J1 DiagnosticsFirewall
Firmware Upgrade by HTTP/TFTPIP Blocklist
NAT TraversalT.38 fax
SIP Trunk SupportPacket Capture Tool
Trunk GroupSIP Peer Mode
SIP Registrar for IP phonesSystem Logs
Web-based ConfigurationMany more

Technical Specifications

Technical Specifications

Technical Specification :
Main Model No.CM300G-E1CM300G-2E1
Sessions3060
Routing FeaturesCall routing and translation (from PCM to IP or reversely)
IP InterfacesDual redundant 2 *100 Base-T Ethernet for VoIP payload and signaling
IP protocolsTCP/UDP, HTTP, ARP/RARP, DNS, NTP, TFTP, TELNET, STUN, etc.
Coder SupportG.711A,G.711U, G.729
Power SupplySingle
TDM Signaling ProtocolsISDN PRI/MF R2/SS7 ISUP/SS7 MTP1~3/SS7 SIGTRAN/SS7 TCAP
Power Requirements12V DC
MountingDesktop
CompatibilityInteroperable with most IP-PBX and UC Platform, and field-proven by SMB and Carriers Worldwide
Dimensions H*W*D(mm)30*190*120
EnvironmentOperating 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
RoutingCall Routing and translation (PCM↔IP)
SafetyCompliant 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/EMICompliant with most international standards. For compliance documents, please contact sales representatives.
OAM&PNetwork Time Protocol(NTP) Web User interface (WebUI) supports configuration via browser SNMP MIBs
Dedicated DSP-Empowered CapabilityTelecom-style DSP algorithm has been optimized for over decades, assuring seamless compliance with any network environment. Plentiful DSP resources are allocated for signaling, media processing, bandwidth optimization, Telco redundancy
Highly Adjustable For Diverse SoftSwitchHomegrown core technologies to assure seamless compliance with diverse softswitch platform Including Mitel, Avaya, Broadsoft, Yate, OpenSIP, Asterisk, VECTRA, VSC, SIPPULSE, Tropico, FreeSwitch and more other softswitch

ONE SOLUTION FOR EVERY INDUSTRY.

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