1 PORT PRI VoIP GATEWAY
- Model: CASQ L301 a single-port PRI VoIP Gateway.
- Design: Engineered for high-quality voice communications over IP networks.
- Functionality: Supports one Primary Rate Interface (PRI) connection for seamless integration between PSTN and VoIP networks.
- Connectivity: Optimized for SIP protocol, ensuring compatibility with various VoIP systems.
- Ideal for: Small to medium-sized businesses requiring reliable voice service over IP infrastructure.





SIP Support
SIP Trunk
OpenVPN
HD Voice Quality
IPv4, UDP/TCP, DHCP
Support Fax T.38/T.30
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Specifications
L301 series wireless gateways which can be compatible with a series of modules (2G/3G/4G), enabling interconnection between GSM / WCDMA / LTE network and VoIP network safely and efficiently. It can bring you excellent HD voice service with multiple codecs, including G.711U, G.711A, GSM, G.722, G.726, G.729. Our products support SMS messages sending, receiving, group sending, and SMS to E-mail.
The L301 series gateways are perfectly compatible with Asterisk, 3CX, FreePBX, FreeSWITCH SIP server, and VOS VoIP system platform. It provides users with diverse telecommunications access methods and helps users reduce communication costs.
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 IPPBX
- 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
- e1/t1 L301
- 2 10/100 Mbps Ethernet ports
- Maximum Power Consumption: 3W
- Power supply specification: 12V/1A
- Operation humidity: 10%~90% non-condensing
- Operating temperature: 0℃~50℃
- Storage temperature: -20℃~70℃
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Features
VoIP Features
- Support SIP, IAX2 Protocol
- Add, Modify & Delete SIP/IAX2 Trunk
- SIP/IAX2 Registration with Domain
- SIP V2.0 RFC3261 Compliance
- DTMF Mode: RFC2833/Inband/SIP Info
- Multiple SIP/IAX2 Registrations modes
- Abundant Codecs: G.711A, G.711U, G.729, G.722, G.726, GSM
Network
- IPv4, UDP/TCP, DHCP, TELNET, HTTP/HTTPS, TFTP
- PPTP VPN (Not available for EAEU)
- HTTP/SSH (Optical Telnet)
- Ping & Traceroute Command on the Web
- Simple Security Strategy: white list, black list, security rules
Management
- Simple and convenient configuration via Web GUI
- Support maintenance and configuration by SSH
- Support configuration files backup and upload
- Support Chinese and English page
- Firmware Update by HTTP
- Support Web and SSH login password modification
- Restore Factory Settings
- CDR (More than 200,000 Lines CDRs Storage Locally)
- System log
- SIP/IAX2 log
- TCP and SIP capture
System Features
- Combine Different SIP/IAX2 Trunk into Group
- CLID Display & Hide (Need operator’s support)
- Random call interval
- Call Duration Limitation
- Single Call Duration Limitation
- Real Open API Protocol (based on Asterisk)
- Support DISA
- SMSC/SMS/USSD
- PIN Identification
- Optional Voice Codec
- Ports Group Management
- SMS Remotely Controlling Gateway
- SMS Bulk Transceiver, Sent to Email and Automatically Resend
- SMS Coding/Detecting Automatically Identification
- SMS Forwarding and Quick Reply
- USSD transceiver
- Outbound
- Automatically Reboot
- Support MMP
- Support for custom scripts, dialplans
- Support cloud management
Certification
- CE
