Unified Communication & Collaboration Solution UCM6301 series

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Unified Communication & Collaboration Solution

UCM6301 series

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.

Supports up to 3000 users and up to 450 concurrent  calls

Zero configuration provisioning of  Grandstream  SIP endpoints

Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints

Wave App allows communication with all UCM6300 users & solutions


and PMS platforms

certificate and random default password to protect calls and accounts

PoE+ and support NAT router

secure  remote connections


VP8 video codec, jitter resilience

up to 50% packet loss


operating system





Analog Telephone FXS Ports

1 RJ11 Port

2 RJ11 Ports

4 RJ11 Ports

8 RJ11 Ports

All ports have lifeline capability in case of power outage; number of ports can be expanded by peering with an FXS gateway

PSTN Line FXO Ports

1 RJ11 Port

2 RJ11 Ports

4 RJ11 Ports

8 RJ11 Ports

All ports have lifeline capability in case of power outage; number of ports can be expanded by peering with an FXO gateway

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


2*USB 3.0, 1*SD card interface

LED Indicators


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

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

Video Codecs

H.264, H.263, H263+, VP8


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, RFC2833, 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


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

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: 715g; Package  Weight: 1211g

Unit Weight: 725g; Package  Weight: 1221g

Unit Weight: 2490g; Package  Weight: 3260g

Unit Weight: 2550g; Package  Weight: 3320g

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/ work-load, in-queue announcement

Customizable Auto Attendant

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

Maximum Call Capacity

Users: 500

Concurrent calls (G.711): 75

Max concurrent SRTP calls (G.711): 50

Users: 1000

Concurrent calls (G.711): 150

Max concurrent SRTP calls (G.711): 100

Users: 2000

Concurrent calls (G.711): 300

Max concurrent SRTP calls (G.711): 200

Users: 3000

Concurrent calls (G.711): 450 Max concurrent SRTP calls (G.711): 300

Maximum Attendees of Conference Bridges

4 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)

Voice Conference: Up to 75 parties (G.711)

6 Video Conference rooms and up to 30 parties with 1080p,

assuming 4 video feeds + 1 screen sharing (H.264 & Opus)

Voice Conference: Up to 150 parties (G.711)

8 Video Conference rooms and up to 60 parties with 1080p,

assuming 4 video feeds + 1 screen sharing (H.264 & Opus)

Voice Conference: Up to 200 parties (G.711)

10 Video Conference rooms and up to 80 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)

Voice Conference: Up to 300 parties (G.711)

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/conferences, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 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 conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to fax

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, ETSI 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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