OpenVox DGW-100X(R) Series E1T1PRI VoIP Gateway

Discription

OpenVox T1/E1 Gateway with open source Asterisk-based VoIP Gateway solution is an operators and call centers. It is a converged media gateway product. This type of gateway connects traditional phone system to IP networks and integrates VoIP PBX and ISDN seamlessly. With user friendly GUI, user may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface). The DGW-100X(R) digital VoIP gateway helps1/2/4 software-selectable T1/E1/PRI interface and helps up to 30/60/120 concurrent calls. The "R" means that the device helps 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 RJ-48
• 2 10/100/1000M Ethernet ports
• 2 USB 2.0 ports
• DGW-100XR with redundant power supply
• Maximum Power Consumption: 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

• Available in 1/2/4 port T1/E1, energy efficiency concurrent processing, upto 120
• Signal ling: PRI/R2/SS7
• Support up to 24 countries’ standard R2 signal ling
• 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
• SIP(RFC3261) compliance
• Support diverse SIP, IAX2 protocols
• Multiple SIP/IAX2 Registration models: None, server, client DTMF Signaling support: RFC2833, In band, SIPINFO
• Real Open API Protocol(based on Asterisk)
• Support static IP and DHCP
• Call Duration Limit
• CLIR(Call Line Identification Restriction)
• Call Waiting
• Call Forwarding (unconditional, no reply, busy, not reachable)
• SMSC/SMS/USSD
• SMS bulk transceiver, SMS automatically resend, Sent to Email
• Support HTTP SMS interface
• SMS Forwarding
• USSD transceiver
• SMS remotely controlling gateway
• IMEI number automatically modify
• PIN identification
• Mobile number portability (MNP)
• Flexible routing settings & unlimited routing rules
• Support DISA
• Extensible Automatic Callback and Speed Dial
• Customizable IVR
• CDR(More than 200,000 Lines CDRs Storage Locally)
• Support configuration files backup and upload
• Support for custom scripts, dial plans
• Least Cost Routing(LCR),according to Time, Port, Calling Number/li>
• Independent System for Each Module
• Affordable Price with Superior Performance
• Hot-swap Design for both Sim cards and Modules
• Compatible with sip server, such as: Asterisk, Elastix, 3CX, Free SWITCH, etc.
• Available for OEM
• 3-Month "No Questions Asked" Return Policy
• One Year Warranty