The UCM6300 audio series enables businesses to create powerful, scalable unified communications and collaboration solutions.
This series of IP PBXs provides a platform that unifies your core 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 1500 users and includes an integrated instant messaging (IM) platform, voice/web conferencing, and the free Wave app that allows users to communicate and collaborate from desktops, mobile devices, IP phones and other devices . SIP endpoints.
It supports the UCM RemoteConnect cloud service for remote users to deliver a best-in-class hybrid platform that combines the control of a local IP PBX with the remote access and system manageability of a cloud solution.
Offering a high-end unified communications and collaboration solution featuring a suite of mobility, security, instant messaging, voice conferencing and collaboration tools, the UCM6300 Audio Series provides a powerful business communications platform for any organization. Characteristics
Supports up to 1500 users and up to 200 simultaneous calls Configuration-free provisioning of Grandstream SIP endpoints Integrated instant messaging (IM), audio conferencing, and web meeting platform that supports access from computers, mobile devices, and SIP endpoints The free Wave app enables simple voice and instant messaging (IM) communications using desktop, 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 your calls and accounts Three Gigabit auto-sensing RJ45 network ports with built-in PoE+ and NAT router support The automated NAT firewall traversal service facilitates secure remote connections Improved reliability with support for Hot Standby High-Availability and dual local deployment Supports full-band Opus voice codec, jitter resilience 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