OpenVox DGW-L202
OpenVox DGW-L202 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 systems to IP networks and integrates VoIP PBX with the ISDN seamlessly. With a friendly GUI, users may easily setup their customized Gateway. Also, secondary development can be completed through AMI (Asterisk Management Interface). The DGW-L202 T1/E1 Gateway supports 2 software-selectable T1/E1 interfaces and supports up to 120 concurrent calls.
Overview
- Product Model: DGW-L202
- T1/E1 Ports: 2
- Concurrent Calls: 60
- Ethernet Ports: 2(10/100/1000M)
- USB: 2(USB 2.0)
- VGA: 1
- Weight: 1352g
- Dimension (W/D/H): 310*162.50*44mm
- Maximum Power Consumption: 18W
- Power Supply: 100-240V/AC
- Operating Temperature: 0?~70?
- Operation Humidity: 5%~95% non-condensing
- Storage Temperature: -40?~85?
Target Applications
- legacy PBX systems to low-cost VoIP services
- legacy PBX systems to remote sites over private VoIP links
- IP PBX systems to legacy TDM services
- The 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
- 2 T1/E1 RJ-48
- 2 10/100/1000Mbps Ethernet ports
- Operation humidity: 5%~95% non-condensing
- Operating temperature: 0?~70?
- 2 USB 2.0 ports
- Maximum Power Consumption: 18W
- Power supply specification: 100-240V/AC
- Storage temperature: -40?~85?
System Features
- 2 port T1/E1, up to 60 energy efficiency concurrent processing
- Signaling: PRI/R2/SS7
- Support up to 24 countries? standard R2 signaling
- 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
- protocols: SIP, IAX, TCP, UDP, RTP, SSH, HTTP, HTTPS
- NTP time synchronization and client time synchronization
- SSH access for background management, Asterisk CLI command operation
- Open API interface (AMI)
- Support ports group management
- Support for custom dial plans
- Firmware update by HTTP
- call statistics
- auto-provision
- channel status show dynamically
- backup/upload configuration file
- Multiple detailed log output
- Support the Chinese language
- Automatically reboot
- Good compatibility, support Asterisk, Elastix, FreeSWITCH, and Small and medium IPPBX platform
- Available for OEM
- 3-month No Question Asked Return Policy
- 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, This gateway registers with the endpoint
- SIP accounts can be registered to multiple servers
- Combine different SIP Trunks into a 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
- DDNS
- Support ping & traceroute command on the web
- network capture on the web
If interested in this product, you can visit the datasheet for more information.
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