VoIP Gateways

DGW-L20X Series E1/T1/PRI Gateway

Name: DGW-L20X Series E1/T1/PRI Gateway
Brand: OpenVox

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

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Product details

OpenVox's DGW-L20X T1/E1 Gateway is an Asterisk-based open-source VoIP gateway solution for carriers and call centers. It is a converged media gateway product. This type of gateway connects traditional telephone systems to IP networks and seamlessly integrates VoIP PBX with ISDN. With a user-friendly graphical interface, users can easily configure their customized gateway. Secondary development can also be performed via AMI (Asterisk Management Interface).

The DGW-L20X T1/E1 Gateway supports 1/2/4 software-selectable T1/E1/PRI interfaces and enables up to 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 different codecs

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

Technical specifications

1/2/4 T1/E1 RJ45

2 10/100/1000Mbps Ethernet ports

2 USB 2.0 ports

1 VGA port

Maximum power consumption: 18W

Power specification: 12V/1A

Operating humidity: 10 %~90 % non-condensing

Operating temperature: 0℃~70℃

Storage temperature: -40℃~85℃

Key features

System features

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

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.723.1, G.722, GSM

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

Supports NTP synchronization 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

Firmware update via HTTP

Supports call statistics

Supports TR069

Supports SNMP

IAX trunk and encrypted transmission support

Automatic provisioning support

Dynamic channel status display

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

Support for adding, editing and deleting SIP accounts

SIP registration with domain

Support for 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

Support for T.38 / pass-through fax

Routing

Flexible routing settings

Support for 512 routes

Support for caller/callee manipulation and filtering

Trunk group support, trunk priority management

Support for adding, editing and deleting routes

E1/T1 port grouping

Failover support

Network Features

Network type: Static IP and DHCP

IPv4, UDP/TCP, DHCP, TFTP, SCP

HTTP/HTTPS/SSH

DDNS support

Support for ping & traceroute commands via the web

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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