Grandstream UCM6301 – IP PBX up to 500 users, 75 calls, 2 video rooms

Grandstream UCM6301 – IP PBX up to 500 users, 75 calls, 2 video rooms

The UCM6300 Series enables businesses to build powerful and scalable unified communications and collaboration solutions. This IP PBX series provides a platform that unifies all business communications into one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms, and more. The UCM6300 Series supports up to 3,000 users and includes a built-in video conferencing and web meeting solution that enables employees to connect from desktop, mobile, GVC Series devices, and IP phones. It can be combined with the UCM6300 ecosystem to deliver a hybrid platform that combines the control of an on-premises IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave web and mobile app, providing a hub for remote collaboration, and UCM RemoteConnect, a cloud-based NAT traversal service to ensure secure remote connections. The UCM6300 Series also offers cloud configuration and management via GDMS and an API for integration with third-party platforms. By offering a high-end unified communications and collaboration solution with a suite of mobility, security, meetings, and collaboration tools, the UCM6300 Series provides a powerful platform for any organization.

Characteristics

• Zero-Configuration Provisioning of Grandstream SIP Endpoints
• Built-in conference and meeting platform; supports desktop endpoints, Wave and SIP applications
• Wave browsers for Android, iOS, Chrome and Firefox enable communication with all UCM6300 users and solutions
• API available for third-party integrations including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with built-in PoE and NAT router support
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports full-band Opus voice codec and H.264/H.263/H.263/H.265/VP8 video codec, jitter resistance up to 50% packet loss
• Compatible with GDMS for cloud configuration, management and monitoring
• Based on the open source telephony operating system Asterisk* version 16