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 organization.

Features

·         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

UCM6300A :

Support Max 250 SIP Users ,Max 50 Concurrent calls

Max 50 concurrent SRTP calls ,200 SIP trunk

Support 3 meeting rooms and up to 50 parties

Without FXO/FXS port , 1*USB 3.0, 1*SD card interface

UCM6302A:

Support Max 500 SIP Users ,Max 75 Concurrent calls 

Max 75 concurrent SRTP calls ,200 SIP trunk,

Support 5 meeting rooms and up to 75 parties

With 2FXO+2FXS port , 1*USB 2.0, 1*USB 3.0, 1*SDcard interface

UCM6304A:

Support Max 1000 SIP Users ,Max 150 Concurrent calls

Max 120 concurrent SRTP calls ,200 SIP trunk

Support 7 meeting rooms and up to 120parties

With 4FXO+4FXS port , 2*USB 3.0, 1*SD card interface

UCM6308A:

Support Max 1500 SIP Users ,Max 200 Concurrent calls

Max 150 concurrent SRTP calls ,200 SIP trunk

Support 9 meeting rooms and up to 150parties

With 8FXO+8FXS port , 2*USB 3.0, 1*SD card interface