+91-9997791085
info@casq.in
2 PORT PRI VoIP GATEWAY
- Model: CASQ L202 a 2-port PRI VoIP Gateway.
- Design: Crafted to deliver high-quality voice communication over IP networks.
- Functionality: Supports two Primary Rate Interface (PRI) connections for seamless integration between PSTN and VoIP networks.
- Connectivity: Optimized for SIP protocol, ensuring smooth compatibility with various VoIP systems.
- Ideal for: Medium to large-sized businesses requiring robust voice services over IP infrastructure.
- Management: Offers centralized cloud-based management for scalable and cost-effective communications.
SIP Support
SIP Trunk
OpenVPN
HD Voice Quality
IPv4, UDP/TCP, DHCP
Support Fax T.38/T.30
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Specifications
L202 Gateway is an open-source Asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. This gateway connects traditional telephone systems to IP networks and integrates VoIP PBX with ISDN seamlessly. With a friendly GUI, users can easily set up their customized gateway. Secondary development can also be completed through the AMI (Asterisk Management Interface).
The L202 Gateway supports 1/2/4 software-selectable T1/E1/PRI interfaces and up to 120 concurrent calls, making it ideal for businesses looking for flexible and efficient communication solutions.
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 interfaces
- 2 10/100/1000Mbps Ethernet ports
- 2 USB 2.0 ports
- Maximum Power Consumption: 18W
- Power supply specification: 100-240V/AC
- Operation humidity: 5%~95% non-condensing
- Operating temperature: 0℃~70℃
- Storage temperature: -40℃~85℃
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Features
System Features
- Available in 1/2/4 port T1/E1, up to 120 energy-efficient concurrent processing
- Signalling: PRI/R2/SS7
- Support up to 24 countries’ standard R2 signalling
- 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 synchronization and client time synchronization
- 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 TR069
- Support auto provision
- Support channel status show dynamically
- Support backup/upload configuration file
- Multiple detailed log outputs
- Support Chinese language
- Automatically reboot
- Good compatibility, support Asterisk, Elastix, Freeswitch, and Small and Medium IPPBX platform
- Available for OEM
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
