OpenVox DGW-100X(R) Gateway – Reliable VoIP Integration for Operators & Call Centers

OpenVox DGW-100X(R) Gateway – Reliable VoIP Integration for Operators & Call Centers

The OpenVox DGW-100X(R) T1/E1 Gateway is a powerful, open-source Asterisk-based VoIP solution designed for seamless integration between traditional telephony systems and IP networks. This converged media gateway effortlessly connects VoIP PBX with ISDN, ensuring high compatibility and reliability.


With an intuitive, user-friendly GUI, users can easily configure and customize their gateway. For advanced customization, the system supports secondary development via AMI (Asterisk Management Interface).


The DGW-100X(R) digital VoIP gateway offers 1, 2, or 4 software-selectable T1/E1/PRI interfaces, handling up to 30, 60, or 120 concurrent calls. The “R” model includes a redundant power supply for enhanced reliability, making it an ideal choice for operators, enterprises, and call centers.

The OpenVox DGW-100X(R) T1/E1 Gateway is a powerful, open-source Asterisk-based VoIP solution designed for seamless integration between traditional telephony systems and IP networks. This converged media gateway effortlessly connects VoIP PBX with ISDN, ensuring high compatibility and reliability.


With an intuitive, user-friendly GUI, users can easily configure and customize their gateway. For advanced customization, the system supports secondary development via AMI (Asterisk Management Interface).


The DGW-100X(R) digital VoIP gateway offers 1, 2, or 4 software-selectable T1/E1/PRI interfaces, handling up to 30, 60, or 120 concurrent calls. The “R” model includes a redundant power supply for enhanced reliability, making it an ideal choice for operators, enterprises, and call centers.

Up to 4 x T1 / E1 RJ45

Up to 4 x T1 / E1 RJ45

DGW-L20X

4 port E1 / T1

2 * 10/100/1000 MBPS eth ports

4 * 10/100MBPS eth ports

Target Applications

Target Applications

Connect legacy PBX systems to low-cost VoIP services

Connect legacy PBX systems to low-cost VoIP services

Connect legacy PBX systems to remote sites over private VoIP links

Connect legacy PBX systems to remote sites over private VoIP links

Connect IP PBX systems to legacy TDM services

Connect IP PBX systems to legacy TDM services

Phased transition from legacy PBX to IPPBX

Phased transition from legacy PBX to IPPBX

Connect virtualized systems to legacy TDM services

Connect virtualized systems to legacy TDM services

Transcoding by connecting systems using varying codecs

Transcoding by connecting systems using varying codecs

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

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

Technical Specifications

Technical Specifications

1/2/4 T1/E1 RJ45

1/2/4 T1/E1 RJ45

2 10/100/1000 Mbps Ethernet ports

2 10/100/1000 Mbps Ethernet ports

2 USB 2.0 ports

2 USB 2.0 ports

DGW-100XR with redundant power supply

DGW-100XR with redundant power supply

DGW-100X with single power supply

DGW-100X with single power supply

Maximum Power Consumption: 18W

Maximum Power Consumption: 18W

Power supply specification: 12V/1A

Power supply specification: 12V/1A

Power supply specification: 12V/1A

Power supply specification: 12V/1A

Operation humidity: 10%~90% non-condensing

Operation humidity: 10%~90% non-condensing

Operating temperature: 0℃~70℃

Operating temperature: 0℃~70℃

Storage temperature: -40℃~85℃

Storage temperature: -40℃~85℃

Model Comparison

DGW-1001

DGW-1001R

DGW-1002

DGW-1002R

DGW-1004

DGW-1004R

Power Supply

1

2

1

2

1

2

Port Type

1

2

4

Concurent Calls

30

60

120

MAX Power

20W

Operation Temperature

0°C ~ 70°C

Storage Temperature Range

-40℃~85℃

Features

System Features

System Features

Available in 1 / 2 / 4 port E1/T1, energy efficiency concurrent processing, up to 30

Available in 1 / 2 / 4 port E1/T1, energy efficiency concurrent processing, up to 30

Signalling:PRI/R2/SS7

Signalling:PRI/R2/SS7

Support up to 24 countries’ standard R2 signalling

Support up to 24 countries’ standard R2 signalling

Support new R2 variant

Support new R2 variant

Simple and convenient configuration via Web GUI

Simple and convenient configuration via Web GUI

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

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

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

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

Support NTP time synchronization and client time synchronization

Support NTP time synchronization and client time synchronization

Support SSH access for background management, Asterisk CLI command operation

Support SSH access for background management, Asterisk CLI command operation

Open API interface (AMI)

Open API interface (AMI)

Support ports group management

Support ports group management

Support for custom dialplans

Support for custom dialplans

Firmware update by HTTP

Firmware update by HTTP

Support call statistics

Support call statistics

Support TR069

Support TR069

Support auto provision

Support auto provision

Support channel status show dynamically

Support channel status show dynamically

Support backup/upload configuration file

Support backup/upload configuration file

Multiple detailed log output

Multiple detailed log output

Support Chinese language

Support Chinese language

Auto reboot

Auto reboot

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

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

SIP Features

SIP Features

Support add, modify & delete SIP Accounts

Support add, modify & delete SIP Accounts

SIP registration with Domain

SIP registration with Domain

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

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

SIP accounts can be registered to multiple servers

SIP accounts can be registered to multiple servers

Combine different SIP Trunks into group

Combine different SIP Trunks into group

SIP(RFC3261) compliance

SIP(RFC3261) compliance

DTMF: RFC2833, SIP INFO, INBAND

DTMF: RFC2833, SIP INFO, INBAND

Support T.38 /Pass-through Fax

Support T.38 /Pass-through Fax

Routing

Routing

Flexible routing settings

Flexible routing settings

Support 512 routing

Support 512 routing

Support caller/callee manipulation and filtering

Support caller/callee manipulation and filtering

Trunk group support, Trunk priority management

Trunk group support, Trunk priority management

Support add, modify & delete routing

Support add, modify & delete routing

E1/T1 port grouping

E1/T1 port grouping

Support Failover

Support Failover

Network Features

Network Features

Network type: Static IP and DHCP

Network type: Static IP and DHCP

IPv4, UDP/TCP, DHCP, TFTP, SCP

IPv4, UDP/TCP, DHCP, TFTP, SCP

HTTP/HTTPS/SSH

HTTP/HTTPS/SSH

Support DDNS

Support DDNS

Support ping & traceroute command on the web

Support ping & traceroute command on the web

Support network capture on the web

Support network capture on the web

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Download

Online Manual

Download

Online Manual

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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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
Company
Careers

VFIDELITY INC.

© 2025 VFIDELITY INC. All Rights Reserved.