Unified Communication & Collaboration Solution
UCM6300 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 thecontrol of an on-premise IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of theWave 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 unifiedcommunications 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 andup to 450 concurrent calls

Zero configuration provisioningof Grandstream SIP endpoints

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

Wave App allows
communication with all
UCM6300 users & solutions

UCM
RemoteConnect
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 andaccounts

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 withsupport for Hot Standby
High-Availability

Supports Full-Band Opus voice
codec and H.264/H.263/H.263+/
VP8 video codec, jitter resilience
up to 50% packet loss

Compatible with GDMS for
cloud setup, management andmonitoring

Based on Asterisk* version
16 open source telephony
operating system

www.grandstream.com

Analog Telephone FXS Ports

PSTN Line FXO Ports
Network Interfaces
NAT Router
Peripheral Ports
LED Indicators

UCM6301

UCM6302

UCM6304

UCM6308

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
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
Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+Yes (supports router mode and switch mode)
1*USB 3.0, 1*SD card interface

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

2*USB 3.0, 1*SD card interface

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

LCD Display

320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar

128x32 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
Voice and Fax Codecs
Video Codecs

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 lossOpus, 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
H.264, H.263, H263+, VP8QoS

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

APIFull API available for third-party platform and application integration

Telephony Operating System
DTMF Methods
Provisioning Protocol &Plug-and-PlayNetwork Protocols
Disconnect Methods
Media Encryption
Universal Power SupplyDimensions
WeightTemperature & HumidityMounting

Multi-Language Support

Caller ID
Polarity Reversal/Wink
Call Center
Customizable Auto Attendant

Based on Asterisk version 16
In-band audio, RFC2833, and SIP INFO
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
SIP, TCP/UDP/IP, RTP/RTCP, IAX, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP,Frame Relay (pending), IPv6, OpenVPN®
Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
SRTP, TLS, HTTPS, SSH, 802.1X, ZRTP
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

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-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
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Yes, with enable/disable option upon call establishment and termination
Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/work-load, in-queue announcement
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 ofConference Bridges

Up to 4 simultaneous videoconference rooms, up to 75
simultaneous participants in all
rooms combined, up to 9 video
feeds in all conference rooms

Up to 8 simultaneous videoconference rooms, up to 150
simultaneous participants in all
rooms combined, up to 9 video
feeds in all conference rooms

Up to 15 simultaneous videoconference rooms, up to 200
simultaneous participants in all
rooms combined, up to 9 video
feeds in all conference rooms

Up to 25 simultaneous videoconference rooms, up to 300
simultaneous participants in all
rooms combined, up to 9 video
feeds in all conference rooms

Wave App

Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCMhosted 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

Internet Protocol Standards

Compliance

Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides acentralized interface to provision, manage, monitor and troubleshoot Grandstream products
RFC 3261, RFC 3262, RFC 3263, RFC 3264, RFC 3515, RFC 3311, RFC 4028. RFC 2976, RFC 3842, RFC 3892, RFC 3428, RFC 4733, RFC 4566, RFC 2617,RFC 3856, RFC 3711, RFC 4582, RFC 4583, RFC 5245, RFC 5389, RFC 5766, RFC 6347, RFC 6455, RFC 8860, RFC 4734, RFC 3665, RFC 3323, RFC 3550,RFC6189
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*Asterisk is a Registered Trademark of Digium, Inc.

10.2024 v2

www.grandstream.com