Unified Communication & Collaboration Solution UCM6300A Audio Series

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Unified Communication & Collaboration Solution

UCM6300A Audio Series

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.

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

(IM) communications using desktops, Web, and Android/ iOS devices

and PMS platforms

certificate and random default password to protect calls and accounts

PoE+ and support NAT router

secure  remote connections

Enhanced reliability with support for Hot Standby High- Availability and local dual deploymen

Supports Full-Band Opus voice codec,jitter resilience up to 50% packet loss


operating system





Analog Telephone FXS Ports


2 RJ11 ports

4 RJ11 ports

8 RJ11 ports

All ports have lifeline capability in case of power outage

PSTN Line FXO Ports


2 RJ11 ports

4 RJ11 ports

8 RJ11 ports

All ports have lifeline capability in case of power outage

Network Interfaces

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

NAT Router

Yes (supports router mode and switch mode)

Peripheral Ports

1*USB 3.0, 1*SD card interface

1*USB 2.0, 1*USB 3.0, 1*SD

card interface

2*USB 3.0, 1*SD card interface

LED Indicators


Power 1/2, FXS, FXO, LAN, WAN,


LCD Display

320×240 color LCD with touch screen for Shortcut Keys and Scroll Bar

128×32 dot matrix graphic LCD with DOWN and OK buttons

Reset Switch

Yes, long press for factory reset and short press for reboot

Voice-over-Packet Capabilities

LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss

Voice and Fax Codecs

Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38


Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS


Full API available for third-party platform and application  integration

Telephony Operating System

Based on Asterisk version 16

DTMF Methods

In-band audio, RFC4733, and SIP INFO

Provisioning Protocol &


Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk

Network Protocols


HDLC, HDLC-ETH, PPP, Frame Relay (pending), IPv6, OpenVPN®

Disconnect Methods

Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect

Media Encryption


Universal Power Supply

Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A

1x DC 12V Power Jack Input: 100~240VAC,

50/60Hz;Output: DC 12V, 2A

2x DC 12V Power Jack Input: 100~240VAC,

50/60Hz;Output: DC 12V, 2A


270mm(L) x 175mm(W) x 36mm(H)

485mm(L) x 187.2mm(W) x 46.2mm(H)


Unit Weight: 705g; Package Weight: 1131g

Unit Weight: 725g; Package Weight: 1221g

Unit Weight: 775g; Package Weight: 1621g

Unit Weight: 2538g; Package Weight: 3463g

Temperature & Humidity

Operating: 32 – 113ºF / 0 ~ 45ºC, Humidity 10 – 90% (non-condensing)

Storage: 14 – 140ºF / -10 ~ 60ºC, Humidity 10 – 90% (non-condensing)


Wall mount & Desktop

Rack mount & Desktop

Multi-Language  Support

-Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish

-Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands

-Customizable language pack to support any other languages

Caller ID

Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT

Polarity Reversal/Wink

Yes, with enable/disable option upon call establishment and  termination

Call Center

Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ workload, in-queue announcement

Customizable  Auto Attendant

Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Maximum Call Capacity

Users: 250

Concurrent calls (G.711): 50 Max concurrent SRTP calls (G.711): 50

Users: 500

Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 75

Users: 1000

Concurrent calls (G.711): 150 Max concurrent SRTP calls (G.711): 120

Users: 1500

Concurrent calls (G.711): 200 Max concurrent SRTP calls (G.711): 150

Maximum Attendees of Conference Bridges

3 meeting rooms and up to 50


5 meeting rooms and up to 75


7 meeting rooms and up to 120


9 meeting rooms and up to 150


Wave App

Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCM-hosted meetings, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 Audio series IP PBX

Call Features

Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice meeting, eventlist, feature codes, busy camp-on/ call completion, voice control

Firmware Upgrade

Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides a centralized interface to provision, manage, monitor and troubleshoot Grandstream   products


FCC: Part 15 (CFR 47) Class B, Part 68

CE: EN 55032, EN 55035, EN 61000-3-2, EN 61000-3-3, EN 62368.1, ES 203 021, ITU-T K.21 IC: ICES-003, CS-03 Part I Issue 9

RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2

Power adapter: UL 60950-1 or UL 62368-1


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