Grandstream UCM6302A Audio IP PBX

The Grandstream UCM6302A IP PBX offers a groundbreaking platform that unifies all essential business communications into a centralized network. From voice and instant messaging to web meetings and facility access, this solution empowers businesses to streamline operations and enhance collaboration. With the UCM6302A, companies can easily manage communication across multiple channels

Advanced Technical Capabilities

Built to support up to 1,500 users and 200 concurrent calls, the UCM6302A provides scalability and robust performance for growing businesses. The system offers zero-configuration provisioning for Grandstream SIP endpoints and includes a full suite of communication tools like instant messaging, voice conferencing, and web meetings. With integrated APIs for CRM and PMS platforms, the UCM6302A enables seamless third-party integrations. Enhanced security features, including secure boot and random default passwords, ensure that communication remains protected at all times.

Streamlined Design

Its three Gigabit auto-sensing RJ45 network ports, complete with integrated PoE+, facilitate quick installations and secure remote connections with automated NAT firewall traversal. The free Wave app, compatible with desktops and mobile devices, provides easy access to voice and messaging. The platform’s support for Hot Standby High-Availability ensures that communication systems remain operational even in critical moments.

Built for Modern Business Needs

Backed by the powerful Grandstream Device Management System (GDMS), this IP PBX simplifies cloud-based management, offering secure and centralized control. With features like full-band Opus codec support and high-level security, it’s the ideal solution for businesses looking to unify, simplify, and future-proof their communication systems.

Grandstream UCM6302A Key Features:

• Handles 1500 users and 200 concurrent calls
• Zero-touch provisioning for Grandstream SIP devices
• Built-in IM, audio conferencing, and web meetings for computers, mobile, and SIP devices
• Free Wave App for desktop, web, and mobile communications
• API support for CRM/PMS integration
• Secure boot, unique certificates, and random default passwords for security
• Three Gigabit RJ45 ports with PoE+ and NAT router
• Automated NAT firewall traversal for secure remote access
• Hot Standby High-Availability and local dual deployment support
• Full-Band Opus codec with 50% packet loss resilience
• GDMS compatible for cloud setup and management
• Based on Asterisk 16 operating system

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

• 2 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 2.0
• 1*USB 3.0
• 1*SDcard 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

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, RFC4733, 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), event list 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
• 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: 500
• Concurrent calls (G.711): 75
• Max concurrent SRTP calls(G.711): 75

Maximum Attendees Of Conference Bridges

• 5 meeting rooms and up to 75 parties

Wave Mobile 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, event list, 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, 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

Datasheet

• Datasheet_UCM6300_Series_English