Grandstream UCM6304A Audio Series IP PBX

The Grandstream UCM6304A Audio VoIP PBX is a cutting-edge communication solution engineered to enhance efficiency and connectivity within your business environment. The UCM6304A features four FXO ports and two FXS ports, offering ample capacity for the connectivity of numerous telephone devices. The product’s 2 RJ45 10/100/1000Mbps ports with integrated PoE Plus are designed to provide swift, efficient data transfer and simplified device installation.

At the core of the UCM6304A is its Zero Configuration provisioning, which allows for quick and effortless setup of Grandstream SIP endpoints. It also supports up to 1000 SIP endpoint registrations, 150 concurrent calls, and up to a maximum of 6 conference bridges with up to 32 simultaneous call participants. Another noteworthy aspect of the UCM6304A is its dual Gigabit RJ45 ports with an integrated NAT router for swift, efficient networking and internet access.

It also comes with three conference bridges that allow for five-way voice conferencing sessions – perfect for collaborative work environments. With security as a priority, the Grandstream UCM6304A also comes equipped with advanced defense mechanisms such as secure boot, unique security certificate per device, random default password per device, encrypted data storage, TLS/SRTP/HTTPS encryption and many more.

Key Features of the UCM6304A Audio Series:

• Compatible with GDMS for easy cloud-based setup, monitoring, and management.
• No monthly per-seat fees—an all-inclusive telephone system.
• Supports up to 1,000 users and 150 concurrent calls.
• Advanced security features, including secure boot, unique certificates, and random default passwords to safeguard calls and accounts.
• Built-in platform for instant messaging, audio conferencing, and web meetings, accessible via computers, mobile devices, and SIP endpoints.
• Zero configuration provisioning for Grandstream SIP endpoints simplifies setup.
• Offers three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and NAT router support.
• Supports Full-Band Opus voice codec with jitter resilience of up to 50% packet loss.
• Includes a free Wave App for voice and instant messaging on desktops, web, and Android/iOS devices.
• API available for third-party CRM and PMS integrations.
• Enhanced reliability with Hot Standby High-Availability and local dual deployment support.
• Automated NAT firewall traversal ensures secure remote connections.
• Based on Asterisk* version 16, a powerful open-source telephony operating system.