Grandstream Ucm6304A Audio Series Ippbx

Description

GRANDSTREAM UCM6304A AUDIO SERIES IPPBX

The Grandstream 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.

FEATURES

Supports up to 1000 users and up to 150 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

Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/iOS devices

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 and accounts

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 with support for Hot Standby High-Availability and local dual deployment

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

Compatible with GDMS for cloud setup, management and monitoring

Based on Asterisk* version 16 open source telephony operating system

SPECIFICATIONS

Manufacturer’s Product Code: UCM6304A

Analog Telephone FXS Ports: 4 RJ11 ports (All ports have lifeline capability in case of power outage)

PSTN Line FXO Ports: 4 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: 2*USB 3.0, 1*SD card interface

LED Indicators: None

LCD Display: 320 x 240 color LCD with touch screen for Shortcut Keys and Scroll Bar

Reset Switch: Yes, long press for factory reset and short press for reboot

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

Provisioning Protocol & Plug-and-Play: Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via

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: 1 x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A

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

Weight: Unit Weight: 775g; Package Weight: 1621g

Mounting: Wall mount & Desktop

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

Customizable 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): 120

Maximum Attendees of Conference Bridges: 7 meeting rooms and up to 120 parties

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

PACKAGE CONTENTS

1 x Grandstream Ucm6304A Audio Series Ippbx