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openvox-gsm-voip-gateways

OpenVox DGW-L20X Series E1T1PRI VoIP Gateway

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

OpenVox DGW-L20X T1/E1 Gateway with open-source Asterisk-based VoIP Gateway solution is an operator and call center. It is a converged media gateway product. This type of gateway connects traditional phone systems to IP networks and smoothly incorporates VoIP PBX and ISDN. With a friendly GUI, users may easily set up their customized Gateway. Also, secondary development can be through AMI (Asterisk Management Interface). The DGW-L20X T1/E1 Gateway helps 1/2/4 software-selectable T1/E1/PRI interface manage up to 120 continuous calls.

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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