1 Port PRI VoIP Gateway
The CASQ L301 is a single-port PRI VoIP Gateway designed to enable high-quality voice communications over IP networks. With support for one Primary Rate Interface (PRI) connection, this gateway provides seamless integration between traditional PSTN networks and modern VoIP platforms. Optimized for reliable and efficient performance, the L301 is ideal for small to medium-sized businesses that require dependable voice service over an IP infrastructure. It supports SIP protocol, allowing easy integration with various VoIP systems, and offers centralized management capabilities through the Cloud, making it an excellent choice for scalable, cost-effective communication solutions.
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What is a 1 Port PRI VoIP Gateway?
Enhancing Communication Systems with a 1 Port PRI VoIP Gateway: The 1 Port PRI VoIP
Gateway is a specialized telecommunications device that bridges traditional telephony with modern
IP-based networks. It allows for the conversion of PRI (Primary Rate Interface) lines to VoIP (Voice
over IP), enabling businesses to connect their legacy PBX systems to IP-based communication
solutions.
Designed for organizations that want to transition to VoIP without replacing their
entire telephony infrastructure, the 1 Port PRI VoIP Gateway offers a cost-effective and scalable
solution. By supporting a single PRI connection, this gateway is ideal for small to medium-sized
businesses that need a manageable yet efficient way to integrate VoIP capabilities with their
existing PRI-based systems.
The 1 Port PRI VoIP Gateway provides high-quality voice
transmission, easy integration with SIP-based systems, and flexibility in routing calls over IP
networks. With this gateway, businesses can optimize their telephony costs, enhance voice quality,
and benefit from the scalability that VoIP technology provides. It also enables organizations to
stay connected across multiple locations, leverage unified communications, and improve call
management efficiency.
How a 1 Port PRI VoIP Gateway Works:
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PRI to VoIP Conversion:
- The 1 Port PRI VoIP Gateway converts traditional PRI voice signals to VoIP (Voice over IP) signals, allowing calls to be transmitted over IP-based networks such as LAN, WAN, or the internet.
- Incoming VoIP calls are also converted back to PRI format, enabling seamless communication with PRI-based systems.
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Single Channel PRI Connection:
- With one dedicated PRI port, the device supports up to 30 simultaneous calls (depending on PRI standards in use), making it suitable for small to medium-sized businesses that rely on PRI for call management.
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PBX and SIP Integration:
- The 1 Port PRI VoIP Gateway integrates smoothly with PBX systems, allowing businesses to route PRI calls through their existing VoIP or SIP-based systems.
- This gateway is ideal for organizations that want to connect PRI lines to IP-PBX systems, providing a scalable solution for modernizing business communications.
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Cost-Effective Communication:
- By utilizing IP networks, businesses can reduce communication costs by routing calls over the more cost-effective internet or local networks instead of traditional phone lines.
- Many 1 Port PRI VoIP Gateways feature least-cost routing, selecting the most affordable path for outgoing calls based on time, destination, and network preferences.
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Enhanced Call Management:
- The gateway provides features like call forwarding, call routing, and monitoring capabilities, enhancing the way businesses manage and control their telephony systems.
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Use Cases:
- Small to Medium Businesses: Transition to VoIP without major infrastructure changes, reducing telephony costs while maintaining PRI connectivity.
- Branch Offices: Connect remote or branch locations with headquarters using VoIP, enabling centralized communications.
- Legacy Systems: Continue to use PRI-based PBX systems while gaining access to VoIP features for flexible communication options.
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Key Components:
- PRI Port: Single PRI port allowing connection of a PRI line with support for multiple simultaneous calls.
- Ethernet Port: Used to connect the gateway to a local IP network, facilitating VoIP communication.
- Control Interface: Typically a web-based interface for configuring call routing, PRI settings, and network preferences.
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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
SIP Support
SIP Trunk
OpenVPN
HD Voice Quality
IPv4, UDP/TCP, DHCP
Support Fax T.38/T.30