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OpenVox ET200X

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 & trace route command on the web
  • Support network capture on the web

Discription

ET200X Gateway is an open source asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. With user friendly GUI, customers may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface). ‘L’ means that the device has no hardware codec V100 module. The ET200X Gateway Helps 1/2/4 software-selectable T1/E1/PRI interface and Helps up to 30/60/120 concurrent calls.

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
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