Grandstream Audio IP PBX voice system support 8 FXO Port , 8 FXS Port

الوصف

Description
The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on 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 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for desktop, web, and mobile, which provides a hub for collaborting remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-party platforms. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300 series provides a powerful platform for any organization.

Features
Supports up to 3000 users and up to 450 concurrent calls
Zero configuration provisioning of Grandstream SIP endpointsUCM6300_back
Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & 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 integrated PoE+ and support NAT router
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 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
Datasheet