Grandstream Audio IP PBX voice system support 2 FXO Port , 2 FXS Port

Description

Description
The UCM6300 Audio series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies 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 1500 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App that allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints. It supports UCM RemoteConnect cloud service for remote users to offer a best-in class hybrid platform that combines 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, security, instant messaging, voice conferencing and collaboration tools, the UCM6300 Audio series provides a powerful business communication platform for any organisation.

NO MONTHLY PER-SEAT FEES. This is an all-inclusive telephone system
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