VoIP Gateways

OpenVox DGW-L301 Series E1/T1/PRI VoIP Gateway

Name: OpenVox DGW-L301 Series E1/T1/PRI VoIP Gateway
Brand: OpenVox

An economical version of an E1/T1/PRI gateway; supports 1 software-selectable E1/T1/PRI interface; supports up to 30 simultaneous calls.

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Product details

OpenVox DGW-L301 E1/T1/PRI Gateway is an open-source Asterisk-based VoIP gateway solution for carriers and call centers. It is a converged multimedia gateway product. This type of gateway connects traditional telephone systems to IP networks and seamlessly integrates the VoIP PBX with the PSTN. With a user-friendly graphical interface, users can easily configure their customized gateway. Secondary development can also be carried out via AMI (Asterisk Management Interface).

The DGW-L301 E1/T1/PRI gateway supports 1 software-selectable E1/T1/PRI interface and supports up to 30 simultaneous 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 IPPBX

Connect virtualized systems to legacy TDM services

Transcoding by connecting systems using various codecs

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

Technical Specifications

1 T1/E1 RJ45

2 Ethernet ports, 10/100 Mbps

Maximum power consumption: 3 W

Power specification: 12 V/1 A

Operating humidity: 10 %~90 % non-condensing

Operating temperature: 0 ℃~50 ℃

Storage temperature: -20 ℃~70 ℃

Key features

System features

Available with 1 E1/T1 port, energy-efficient, simultaneous processing, up to 30

Signaling: PRI/R2/SS7

Supports up to 24 countries for standard R2 signaling

Supports the new R2 variant

Simple and convenient configuration via the Web GUI

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

Supports protocols: SIP, IAX, TCP, UDP, RTP, SSH, HTTP, HTTPS

Supports NTP time synchronization and client time synchronization

Supports SSH access for backend management, Asterisk CLI command operation

Open API interface (AMI)

Supports port group management

Supports custom call plans

Firmware update via HTTP

Supports call statistics

TR069 support

Automatic provisioning support

Dynamic display of channel status

Configuration file backup/download

Multiple detailed log outputs

Chinese language support

Automatic restart

Good compatibility, supports Asterisk, Elastix, Freeswitch, and medium-sized IPPBX platforms

Available for OEM

3-month “No Questions Asked” return policy and 2-year warranty

SIP Features

Supports adding, modifying, and deleting SIP accounts

SIP registration with domain

Supports multiple SIP registrations: Anonymous, the endpoint registers with this gateway, 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 / Pass-through Fax

Routing

Flexible routing settings

Supports 512 routes

Supports caller/callee manipulation and filtering

Supports trunk groups and trunk priority management

Supports adding, modifying, 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

DDNS support

Supports ping & traceroute commands on the web

Supports network capture on 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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