OpenVox DGW-L302 Series E1/T1/PRI VoIP Gateway
Supports 2 software-selectable T1/E1/PRI interface;
Supports 60 concurrent calls.
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Specifications
The OpenVox DGW-L302 is a digital VoIP gateway equipped with E1/T1 interfaces, designed to connect ttraditional digital telephone lines (PSTN and ISDN) with VoIP networks. As a converged media gateway, it enables seamless integration of traditional telephone systems into IP networks, achieving perfect convergence between VoIP telephony and conventional telephone infrastructures.
Users can easily configure and customize the gateway through an intuitive and user-friendly graphical interface. Additionally, it supports powerful API interfaces, facilitating secondary development and deep customization.
The DGW-L302 offers 2 software-selectable T1/E1 interfaces, supports PRI, SS7, and R2 signaling, and can handle up to 60 concurrent calls, meeting communication needs of various scales.
DGW-L302 Digital Gateway E1/T1 Ports 2 Concurrent Calls 60 Ethernet Port 2 * 1000Mbps ports Console RS232(RJ45) Weight 1.3kg Dimension 232mm*152mm*45mm Maximum Power 12W Power Supply 12V 1A Operating temperature 0℃~50℃ Operation Humidity Range 10%-90% Non-condensing Storage temperature -20℃~70℃ Certification CE -
Features
Physical Interfaces
- 2 E1/T1 Port
- 1 * RS232(RJ45)
- 1 * WAN: 1000Mbps, RJ-45
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1 * LAN: 1000Mbps, RJ-45
Voice&FAX
- G.711A/U, G.723.1, G.729A, iLBC, OPUS, ARM and ARM-WB
- Comfort Noise Generation(CNG)
- Adaptive (Dynamic) Jitter Buffer
- Programmable Gain Control
- DTMF mode: SIP INFO/RFC4733/INBAND
- T.38/Pass-through
- Echo Cancellation(G.168), with up to 128ms
System Features
- T1/E1/PRI, up to 60 energy efficiency concurrent processing
- Signaling: PRI/R2/SS7
- Support up to 24 countries’ standard R2 signaling
- Simple and convenient configuration via Web GUI
- Support protocols: SIP, TCP, UDP, RTP, SSH, HTTP, HTTPS
- Support NTP time synchronization and client time synchronization
- Support SSH access for background management, Asterisk CLI command operation
- OpenAPI for all functions
- 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 scheduled reboot
- Support Asterisk, Issabel,3CX, Freeswitch , VOS, and BroadSoft IPPBX platforms
PSTN
- ISDN PRI
23B+D(T1),30B+D(E1),NT or TE ITU-T Q.921, ITU-T Q.931, Q.Sig
- Signal 7/SS7
ITU-T, ANSI，ITU-CHINA MTP1/MTP2/MTP3, ISUP
- E1 Frame Type: DF,CRC-4,CRC_ITU
- T1 Frame Type:
2-Frame Multi-frame (F12, D3/4) Extended Super-frame (F24, ESF) - Line Codes: E1:HDB3 T1:B8ZS
- Clock: Local/Remote Clock Source
Network Features
- Network type: Static IP, DHCP
- IPv4, IPv6 UDP/TCP, DHCP, TFTP, SCP
- HTTP/HTTPS/SSH
- Support ping & traceroute command on the web
- Support network capture on the web
- Support MGT and IP aliases
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Layer 2 QoS and Layer 3 QoS
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dnssrv fast switching
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loopback network topology hiding
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Cloud-based Management
Routing
- Flexible routing settings
- Support 512 routing
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Support caller/callee manipulation and filtering
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Support add, modify & delete routing
- E1/T1 port grouping
- Support Failover
VoIP Protocol
- SIP v2.0 (UDP/TCP), RFC326
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SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311 SIP TLS/SRTP
- RTP/RTCP, RFC2198, 1889
- SIP-T,RFC3372, RFC3204, RFC3398
- SIP Trunk Work Mode : Peer/Access
- SIP/IMS Registration :With up to 2000 SIP Accounts
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NAT: Dynamic NAT, Rport
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Download
PDF Manual
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Demo