All stock codes associated to this product
UCM6300A, IPGUCM6300A, IPG-UCM6300A, GRUCM6300A, GR-UCM6300A, 6947273703549
UCM6300A On Premise IP PBX (Audio Only) with No FXO ports, No FXS ports and supporting up to 250 users
The UCM6300 Audio series enables enterprises to create strong and scalable unified communication and collaboration solutions. This IP PBX Audio series provides a platform that integrates basic business communications needs such as voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access, intercoms, and more.
The UCM6300A can support up to 250 users and 50 concurrent calls and has a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App, which allows users to connect and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints.
It offers 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 providing UCM RemoteConnect cloud service for distant users.
The UCM6300A provides a robust business communication platform for any organization by providing a high-end unified communications and collaboration solution packed with a suite of mobility, security, instant messaging, voice conferencing, and collaboration tools.
Features
Supports up to 250 users and up to 50 concurrent calls
Zero configuration provisioning of Grandstream SIP endpoints
Built-in Instant Messaging, Audio Conferencing &Web Meetings platform supporting access from computers, mobile devices, 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
