Unified Communications Platform

DTU-30X (Digital Trunk Unit)

Name: DTU-30X (Digital Trunk Unit)
Brand: OpenVox

Supports 1/2/4 software-selectable T1/E1/PRI interfaces; supports up to 30/60/120 concurrent calls.

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Product details

DTU-30X (Digital Trunk Unit) is an open-source Asterisk-based VoIP gateway solution for operators and call centers. It is a converged multimedia gateway product that can be used with products in the OpenVox UCP series. This type of gateway connects traditional telephone systems to IP networks and seamlessly integrates the VoIP PBX with PRI/SS7/R2. With a user-friendly graphical interface, customers can easily configure their customized gateway. Secondary development can also be performed via the API.

The DTU-301/302/304 supports a software-selectable 1/2/4 T1/E1 interface and supports up to 30/60/120 concurrent calls.

Target applications

Connect legacy PBX systems to low-cost VoIP services

Connect legacy PBX systems to remote sites via private VoIP links

Connect IP PBX systems to legacy TDM services

Gradual transition from legacy PBX to IP PBX

Connect virtualized systems to legacy TDM services

Transcoding by connecting systems that use various codecs

Lync connectivity to SIP or legacy TDM providers and to SIP or legacy PBXs

Technical specifications

1/2/4 E1/T1 RJ45

2 10/100M Ethernet ports (one on the front panel and one on the rear panel)

Power consumption: 3W maximum.

Operating temperature: 0°C to 50°C

Storage temperature: -20°C to 70°C

Operating ambient humidity: 10%-90% non-condensing

Key features

System features

Available in 1/2/4 T1/E1 ports, energy-efficient, concurrent processing, up to 30/60/120 channels

Signaling: PRI/R2/SS7

Supports standard R2 signaling in 24 countries

Supports new R2 variants

Simple and convenient configuration via the Web GUI

Codec support: G.711A, G.711U, G.729A, G.722, GSM

Protocol support: SIP, IAX, TCP, UDP, RTP, SSH, HTTP, HTTPS

Supports NTP and client time synchronization

Supports SSH access for backend management and Asterisk CLI command operations

Open API interface (AMI)

Supports port group management

Supports customized dial plans

G.168 echo cancellation

Firmware update via HTTP

Supports call statistics

Automatic provisioning support

Dynamic channel status display

Configuration file backup/download

Multiple detailed log outputs

Chinese language support

Automatic restart

Compatible with all types of SIP servers, such as Asterisk, 3CX, Freeswitch, and other IPPBX platforms

Available for OEM/ODM

3-month “No Questions Asked” return policy and a two-year warranty

SIP Features

Support for adding, editing and deleting SIP accounts

SIP registration with domain

Supports multiple SIP registrations: Anonymous, the endpoint registers with this gateway, and this gateway registers with the endpoint

SIP accounts can be registered on multiple servers

Combine different SIP trunks into a group

SIP compliance (RFC3261)

DTMF: RFC2833, SIP INFO, INBAND

Supports T.38 / Fax Pass-through

Routing

Flexible routing settings

Support for 512 routes

Supports caller/called number manipulation and filtering

Supports trunk groups and trunk priority management

Support for adding, editing and deleting routes

E1/T1 port grouping

Supports failover

Network Features

Network type: Static IP and DHCP

IPv4, UDP/TCP, DHCP, TFTP, SCP

HTTP/HTTPS/SSH

Supports DDNS

Supports ping & traceroute commands via the web interface

Support for network capture via the web

Specifications

See the manufacturer description and linked documentation for product information. No specification table is published on the captured product webpage.

Documents

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