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Grandstream UCM6302A IP PBX Audio Only


The Grandstream UCM6300A Audio Series brings a reliable on-premise IP PBX solution with the flexibility for remote access.

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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
  • Key Features

    • Up to 500 users
    • Up to 75 concurrent calls
    • 5 meeting rooms
SKU: UCM6302A Categories: ,

System Functionality :

VOICE/AUDIO ONLY CAPABLE

Max. Call Capacity :

75

SIP Extension Capacity :

500

VoIP Trunk Capacity :

75

Analog (FXO) Phone Line Capacity :

2

Analog (FXS) Extension Ports :

2

Gigabit Network Ports:

Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+

Expandable Storage Memory:

Yes – 2x USB + SD Card or powered USB drive

Processor:

4 Core Processor

Internal Memory:

1 GB RAM and 32GB flash

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