WhatsApp Icon

2 Port PRI VoIP Gateway

The CASQ L202 is a 2 port PRI VoIP Gateway crafted to deliver high-quality voice communication over IP networks. With support for two Primary Rate Interface (PRI) connections, this gateway provides a seamless bridge between traditional PSTN networks and modern VoIP platforms. Optimized for reliability and efficiency, the L202 is ideal for medium to large-sized businesses that need robust voice services over an IP infrastructure. Supporting the SIP protocol, it integrates easily with various VoIP systems and offers centralized cloud-based management, making it a versatile, scalable solution for cost-effective communications.

Contact
2 Port PRI VoIP Gateway
2 Port PRI VoIP Gateway 2 Port PRI VoIP Gateway 2 Port PRI VoIP Gateway 2 Port PRI VoIP Gateway

2 port pri gateway

What is a 2 Port PRI VoIP Gateway?

Enhancing Communication Systems with a 2 Port PRI VoIP Gateway: The 2 Port PRI VoIP Gateway is a versatile telecommunications device designed to bridge traditional telephony with IP-based networks. It converts two PRI (Primary Rate Interface) lines to VoIP (Voice over IP), allowing businesses to connect legacy PBX systems to IP-based communication platforms seamlessly.
With support for two PRI connections, this gateway is ideal for medium to large-sized organizations that require higher call capacity and want to transition to VoIP without overhauling their entire telephony setup. The 2 Port PRI VoIP Gateway provides a scalable, cost-effective solution for businesses looking to integrate VoIP capabilities with existing PRI-based systems.
Offering high-quality voice transmission and seamless integration with SIP-based systems, the 2 Port PRI VoIP Gateway enhances call quality, improves telephony cost efficiency, and supports flexible call routing over IP networks. This gateway allows organizations to manage communications across multiple locations, leverage unified communications, and improve call handling efficiency, making it a powerful choice for modernizing business telephony.

How a 2 Port PRI VoIP Gateway Works:

  1. PRI to VoIP Conversion:

    • The 2 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.
  2. Dual Channel PRI Connection:

    • With two dedicated PRI ports, the device supports up to 60 simultaneous calls (depending on PRI standards in use), making it suitable for medium to large-sized businesses that rely on PRI for call management.
  3. PBX and SIP Integration:

    • The 2 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 multiple PRI lines to IP-PBX systems, providing a scalable solution for modernizing business communications.
  4. 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 2 Port PRI VoIP Gateways feature least-cost routing, selecting the most affordable path for outgoing calls based on time, destination, and network preferences.
  5. 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.
  6. Use Cases:

    • Medium to Large 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.
  7. Key Components:

    • PRI Ports: Two PRI ports allowing connection of multiple PRI lines with support for a higher volume of 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.
  • 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℃
  • 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

 
SIP Support

SIP Support

SIP Trunk

SIP Trunk

OpenVPN

OpenVPN

HD Voice Quality

HD Voice Quality

IPv4

IPv4, UDP/TCP, DHCP

Fax T.38/T.30

Support Fax T.38/T.30