Grandstream UCM6302 IP PBX

Grandstream UCM6302 IP PBX allows businesses to build powerful and scalable unified communication and collaboration solutions.

Provides 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 UCM6302 IP PBX 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 Grandstream IP PBX with the remote access of a cloud solution.

The UCM6300 ecosystem consists of the Wave app for web and mobile, which provides a hub for collaborating remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections.

Grandstream UCM6302 PBX 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 UCM6302 provides a powerful platform for any organisation.

Product Specifications

Analog Telephone FXS Ports

• 2 RJ11 Port
• All ports have lifeline capability in case of power outage

PSTN Line FXO Ports

• 2 RJ11 Port
• 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 2.0
• 1*USB 3.0
• 1*SD card interface

LED Indicators

• None

LCD Display

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

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+, H.265, VP8

QoS

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

API

• 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 & Plug-and-Play

• 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

• TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, 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

• SRTP, TLS, HTTPS, SSH, 802.1X

Universal Power Supply

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

Dimensions

• 270mm(L) x 175mm(W) x 36mm(H)

Weight

• Unit Weight: 725g
• Package Weight: 1221g

Temperature & Humidity

• Operating: 32 – 113ºF / 0 ~ 45ºC, Humidity 10 – 90% (non-condensing)
• Storage: 14 – 140ºF / -10 ~ 60ºC, Humidity 10 – 90% (non-condensing)

Mounting

• Wall mount & Desktop

Multi-Language Support

• Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish
• Customisable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands
• Customisable 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

Customisable Auto Attendant

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

Maximum Call Capacity

• Users: 1000
• Concurrent calls (G.711): 150
• Max concurrent SRTP calls (G.711): 100

Maximum Attendees of Conference Bridges

• 3 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & G.711)
• Voice Conference: Up to 150 parties (G.711)

Wave Mobile App

• Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300

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

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

Compliance

• 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

Brand