

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
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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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
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
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
VFIDELITY INC.
© 2025 VFIDELITY INC. All Rights Reserved.