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ISDN / PRI / E1 / T1 TRUNK GATWAY - Aegis XNTEL
SIP-to-PRI Trunk Gateway for Seamless IP & Legacy Telephony Integration
The Aegis XNTEL UCM300D-E1 is a compact and reliable SIP-to-PRI Trunk Gateway designed to connect modern IP-based communication systems with existing ISDN PRI, E1 and T1 telephony infrastructure. It enables organizations to gradually migrate toward SIP-based communication while continuing to use their existing PBX, IP-PBX, EPABX and other legacy telephony systems.
With support for up to 30 simultaneous sessions, flexible call routing and translation between PCM and IP, and support for multiple TDM signaling protocols, the UCM300D-E1 provides a practical solution for integrating traditional digital telephone networks with SIP trunks and IP-based communication platforms.
Its compact 30 × 190 × 120 mm desktop form factor, software-configurable E1/T1 ports, dual redundant Ethernet interfaces and dedicated DSP-based processing make it suitable for enterprise, SMB, carrier and service-provider environments where reliable voice connectivity and controlled migration to IP are required.
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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.
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 1/2 E1 in Mini Size
- Support SIP and ISDN/R2/CAS/ SS7 (Optional) and More
- Up to 60 simultaneous VoIP to ISDN calls
- Superior Voice Quality and Reliability in Full Load
- Only 30mm*190mm *120mm(High*Wide*Deep)
- Cost-effective call routing
- Software configurable E1/T1 Ports
- Connect Airtel, TATA, JIO, BSNL support open VPN ease of connection
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Aegis PRI VoIP Gateway offers a simple and reliable solution to connect an IP-based system to a T1/E1/PRI line. The PRI system undoubtedly represents a communications industry revolution. Your old legacy phone lines can successfully handle data, voice, video, and other types of traffic with the help of Primary Rate Interface (PRI). You can make and receive many calls simultaneously while utilising the PRI voip gateway, in addition. For the purpose of supporting seamless service delivery, it transforms digital E1 PSTN messages into IP formats and secures sessions across IP and mixed network boundaries.
Salient Features
Flexible Scalability
A wide variety of gateway toolkits support gateway maintenance and software upgrades for Web UI, gateway services, and firmware. From 30 to 60 simultaneous SIP connections with multimedia transcoding offers excellent performance in a small footprint to help minimise ownership cost and operational cost.
Potent and Flexible Gateway Solution
Clearly define the migration path to an all-IP network. It offers media transcoding and excellent sessions per second performance while scaling up to 60 simultaneous IP sessions; Support call routing and call automatic failover for outbound routing from IP to TDM.
Any-to-Any Signaling and Transcoding
Through its capacity to integrate several protocols to deliver services, it can provide any-to-any network access. It may offer interworking between SS1, R2, ISDN, SS7 (Optional), SIP formats as well as any-to-any media transcoding for well-known voice codecs, T.38 and G.711 fax interworking, RTP, INBAND, and SIPINFO.
User-Friendly Management & Toolkits
The Web graphical user interface (WebUI) is a real-time web toolkit for configuring, monitoring, and maintaining web applications. Flexible configuration of SIP and protocols makes it easier to set up SIP, SIP trunking, SIP mediation, PCM, SS7 (Optional) and ISDN, routing, and other features.
Combined All Features on A Single Platform
TDM and IP interworking is made possible by integrated multimedia gateway features, which also enable automated failover between domains and flexible service delivery.
Integrated Transcoding for Voice, Tone and Faxing
Reduces CAPEX and the number of platforms deployed by eliminating the need to add additional hardware to fulfil transcoding requirements. It also supports a variety of codecs, including G.723, G.729, G.711, iLBC, SIPINFO, RFC2833, RF3261, INBOUND, and more.
SIP to PRI Trunk VoIP Gateway Features
| Blocklist | Call Duration Limitation |
| Caller ID Prefix | Call Detail Record (CDR) |
| Caller ID | Configure backup/restore |
| Call Routing Rules | Custom Prompts |
| E1/T1/J1 Diagnostics | Firewall |
| Firmware Upgrade by HTTP/TFTP | IP Blocklist |
| NAT Traversal | T.38 fax |
| SIP Trunk Support | Packet Capture Tool |
| Trunk Group | SIP Peer Mode |
| SIP Registrar for IP phones | System Logs |
| Web-based Configuration | Many more |
Technical Specifications
| Technical Specification : | ||
|---|---|---|
| Main Model No. | UCM300D-E1 | UCM300D-2E1 |
| Sessions | 30 | 60 |
| Routing Features | Call routing and translation (from PCM to IP or reversely) | |
| IP Interfaces | Dual redundant 2 *100 Base-T Ethernet for VoIP payload and signaling | |
| IP protocols | TCP/UDP, HTTP, ARP/RARP, DNS, NTP, TFTP, TELNET, STUN, etc. | |
| Coder Support | G.711A,G.711U, G.729 | |
| Power Supply | Single | |
| TDM Signaling Protocols | ISDN PRI/MF R2/SS7 ISUP/SS7 MTP1~3/SS7 SIGTRAN/SS7 TCAP | |
| Power Requirements | 12V DC | |
| Mounting | Desktop | |
| Compatibility | Interoperable with most IP-PBX and UC Platform, and field-proven by SMB and Carriers Worldwide | |
| Dimensions H*W*D(mm) | 30*190*120 | |
| 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 | |
| Routing | Call Routing and translation (PCM↔IP) | |
| 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 | |
| Dedicated DSP-Empowered Capability | Telecom-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 SoftSwitch | Homegrown 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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