GRANDSTREAM | UCM6300 Audio Series IP PBX

The UCM6300 Audio series allows businesses to build powerful an d scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifie s fundamental business communications needs, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 Audio Series supports up to 150 0 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App th at allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoin ts. It supports UCM RemoteConnect cloud service for remote users to offer a best-in-class hybrid platform that c ombines the control of an on-premise IP PBX with the remote access and system manageability of a cloud solution. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, sec urity, instant messaging, voice conferencing and collaboration tools, the UCM6300 Audio series provides a powerf ul business communication platform for any organization.

• Unified Communication & Collaboration Solution.

• Supports up to 1500 users and up to 200 concurrent calls.

• Zero configuration provisioning of Grandstream SIP endpoints.

• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.

• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices.

• 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 integrated PoE+ and support NAT router.

• Automated NAT firewall traversal service facilitates secure remote connections.

• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment.

• Supports Full-Band Opus voice codec,jitter resilience up to 50% packet loss.

• Compatible with GDMS for cloud setup, management and monitoring.

• Based on Asterisk* version 16 open source telephony operating system.