OpenVox DGW1001(R) 1 port E1/T1 Digital VoIP Gateway mit Redundanten Netzteil (Redundant Power); Signalling: PRI/R2/SS7

OpenVox DGW-100XR series 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 PSTN seamlessly. With friendly GUI, users may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface).

There are four E1/T1 Gateway models, the DGW-1001(R), DGW-1002(R), DGW-1004(R) and DGW-1008(R). There is one port on DGW-1001 supporting 30 channels most, two ports on DGW-1002 supporting 60 channels most, four ports on DGW-1004 supporting 120 channels and eight ports on DGW-1008 supporting 240 channels most. The "R" means that the device supports redundant power supply. It is developed with a wide selection of codecs and signaling protocol, including G.711A, G.711U, G.729A, G.722, G.723 and GSM. It supports PRI protocol. OpenVox T1/E1 Gateway has good processing ability and stability and we provides 1/2/4/8 T1/E1 interface for your choice. The DGW-100XR series gateway will be 100% compatible with all kind of SIP servers, such as Asterisk, Askozia, Elastix, trixbox, 3CX, FreeSWITCH and VOS VoIP operating plattform.

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 T1/E1  RJ-48
• 2 10/100/1000M Ethernet ports
• 2 USB 2.0 ports
• DGW-1001R with redundant power supply
• Maximum power: 20W
• Power supply specification: 100-240V/AC
• Operation humidity: 5%~95% non-condensing
• Operating temperature: 0°C~70°C
• Storage temperature: -40°C~85°C

Features

System Features

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

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