DGW-L301 Series

Software-selectable Interface.
Supports 30 concurrent calls.

Powerful, open-source Asterisk®-based VoIP gateway designed for operators and call centers. It bridges traditional telephone systems with IP networks, enabling smooth VoIP PBX and PSTN integration.
Powerful, open-source Asterisk®-based VoIP gateway designed for operators and call centers. It bridges traditional telephone systems with IP networks, enabling smooth VoIP PBX and PSTN integration.

User-friendly GUI for effortless setup and AMI (Asterisk Management Interface).

Software-selectable E1/T1/PRI interface, the DGW-L301 supports up to 30 concurrent calls.

Interface

1 x T1 / E1 / PRI RJ45

Ethernet ports

2 x 10/100 MBPS Ethernet ports

Power consumption

3W maximum power consumption

Applications

Applications

Connect legacy PBX systems to low-cost VoIP services

Connect legacy PBX systems to remote sites over private VoIP links

Connect IP PBX systems to legacy TDM services

Phased transition from legacy PBX to IPPBX

Connect virtualized systems to legacy TDM services

Transcoding by connecting systems using varying codecs

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

Technical Specifications

Technical Specifications

1 T1/E1 RJ45

2 10/100 Mbps Ethernet ports

Maximum Power Consumption: 3W

Power supply specification: 12V/1A

Operation humidity: 10%~90% non-condensing

Operating temperature: 0℃~50℃

Storage temperature: -20℃~70℃

Features

Features

System Features

System Features

Available in 1 port E1/T1, energy efficiency concurrent processing, up to 30

Signalling:PRI/R2/SS7

Support up to 24 countries’ standard R2 signalling

Support new R2 variant

Simple and convenient configuration via Web GUI

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

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

Support NTP time synchronization and client time synchronization

Support SSH access for background management, Asterisk CLI command operation

Open API interface (AMI)

Support ports group management

Support for custom dialplans

Firmware update by HTTP

Support call statistics

Support TR069

Support auto provision

Support channel status show dynamically

Support backup/upload configuration file

Multiple detailed log output

Support Chinese language

Auto reboot

Good compatibility, support Asterisk, Elastix, Freeswitch and Small and medium IPPBX platform

SIP Features

SIP Features

Support add, modify & delete SIP Accounts

SIP registration with Domain

Support multiple SIP registrations: Anonymous, Endpoint registers with this gateway, This gateway registers with the endpoint

SIP accounts can be registered to multiple servers

Combine different SIP Trunks into group

SIP(RFC3261) compliance

DTMF: RFC2833, SIP INFO, INBAND

Support T.38 /Pass-through Fax

Routing

Routing

Flexible routing settings

Support 512 routing

Support caller/callee manipulation and filtering

Trunk group support, Trunk priority management

Support add, modify & delete routing

E1/T1 port grouping

Support Failover

Network Features

Network Features

Network type: Static IP and DHCP

IPv4, UDP/TCP, DHCP, TFTP, SCP

HTTP/HTTPS/SSH

Support DDNS

Support ping & traceroute command on the web

Support network capture on the web

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Office Address:
22 Garnet St. Capitol View Park Subd.
Brgy. Bulihan, Malolos, Bulacan
Phone:
+63 917 700 2984
Viber:

+63 917 700 2984

Email:
hello@voicefidelity.com
Company
Careers

VFIDELITY INC.

© 2025 VFIDELITY INC. All Rights Reserved.

Office Address:

22 Garnet St. Capitol View Park Subd.

Brgy. Bulihan, Malolos, Bulacan

Phone:

+63 917 700 2984

Viber:

+63 917 700 2984

Email:

hello@voicefidelity.com

Get Help

FAQ

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Careers

VFIDELITY INC.

© 2025 VFIDELITY INC.

All Rights Reserved.

Office Address:
22 Garnet St. Capitol View Park Subd.
Brgy. Bulihan, Malolos, Bulacan
Phone:
+63 917 700 2984
Viber:

+63 917 700 2984

Email:
hello@voicefidelity.com
Company
Careers

VFIDELITY INC.

© 2025 VFIDELITY INC. All Rights Reserved.