OpenVox DGW-L201 T1/E1 VoIP Gateway
![OpenVox DGW-L201 T1/E1 VoIP Gateway]()
![OpenVox DGW-L201 T1/E1 VoIP Gateway]()
OpenVox DGW-L201 T1/E1 VoIP Gateway
Available:In Stock
£505.56
OpenVox DGW-L201 T1/E1 VoIP Gateway
The OpenVox DGW-L201 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 friendly GUI, users may easily setup their customised Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface). The DGW-L201 supports 1 software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls.
OpenVox DGW-L201 Key Features
- Seamless Integration
- Friendly GUI
- T1/E1 ports
- 2 10/100/1000M Ethernet Ports
- 2 USB ports
- 100-240V/AC Power Supply
OpenVox DGW-L201 Technical Specification
General
- Product Model: DGW-L201
- T1/E1 Ports: 1
- Concurrent Calls: 30
- 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
- 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 10/100/1000Mbps 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
- Available in 1 port T1/E1,up to 30 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
- Support protocols: SIP, IAX, TCP, UDP, RTP, SSH, HTTP, HTTPS
- Support NTP time synchronisation and client time synchronisation
- Support SSH access for background management, Asterisk CLI command operation
- Open API interface (AMI)
- Support ports group management
- Support for custom dialplans
- 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
- 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 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
OpenVox DGW-L201 T1/E1 VoIP Gateway
The OpenVox DGW-L201 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 friendly GUI, users may easily setup their customised Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface). The DGW-L201 supports 1 software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls.
OpenVox DGW-L201 Key Features
- Seamless Integration
- Friendly GUI
- T1/E1 ports
- 2 10/100/1000M Ethernet Ports
- 2 USB ports
- 100-240V/AC Power Supply
General
- Product Model: DGW-L201
- T1/E1 Ports: 1
- Concurrent Calls: 30
- 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
- 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 10/100/1000Mbps 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
- Available in 1 port T1/E1,up to 30 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
- Support protocols: SIP, IAX, TCP, UDP, RTP, SSH, HTTP, HTTPS
- Support NTP time synchronisation and client time synchronisation
- Support SSH access for background management, Asterisk CLI command operation
- Open API interface (AMI)
- Support ports group management
- Support for custom dialplans
- 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
- 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 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
