OpenVox DGW-100X(R) Series E1/T1/PRI VoIP Gateway
The digital E1/T1 Gateway with up to 4 software-selectable E1/T1/PRI interface;
Supports 30/60/120 concurrent calls.
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Specifications
OpenVox T1/E1 Gateway is an open source asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. This kind of gateway connects traditional telephone system to IP networks and integrates VoIP PBX with the ISDN seamlessly. With user friendly GUI, customers may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface).
The DGW-100X(R) digital VoIP gateway supports 1/2/4 software-selectable T1/E1/PRI interface and supports up to 30/60/120 concurrent calls. The "R" means that the device supports redundant power supply.
Target 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 IP PBX
- 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
- 1/2/4 T1/E1 RJ45
- 2 10/100/1000M Ethernet ports
- 2 USB 2.0 ports
- DGW-100XR with redundant power supply
- DGW-100X with single power supply
- Maximum Power Consumption: 20W
- Power supply specification: 100-240V/AC
- Operation humidity: 10%~90% non-condensing
- Operating temperature: 0℃~70℃
- Storage temperature: -20℃~70℃
DGW-100X(R) E1/T1/PRI VoIP Gateway Product Model DGW-1001 DGW-1001R DGW-1002 DGW-1002R DGW-1004 DGW-1004R Power Supply 1 2 1 2 1 2 Ports 1 2 4 Concurrent Calls 30 60 120 Maximum power 20W Operating temperature 0℃~70℃ Storage temperature -40℃~85℃ -
Features
System Features
- Available in 1/2/4 port T1/E1, energy efficiency concurrent processing, 30/60/120
- 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.723.1, 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 custom dialplans
- Echo Cancellation (Octasic® DSP)
- 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
- Automatically reboot
- Compatibility with all kinds of SIP servers, such as Asterisk, Elastix, Freeswitch and other IPPBX platforms
- Available for OEM
- 3-month "No Question Asked" Return Policy, and Two-year Warranty
SIP Features
- Support add, modify & delete SIP Accounts
- SIP registration with Domain
- Support multiple SIP registrations：Anonymous, Endpoint registers with this gateway and 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
- 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 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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