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Grandstream UCM6302A | IP PBX (Audio only)

Grandstream UCM6302A | IP PBX (Audio only)

Original price was: ₦502,375.20.Current price is: ₦478,900.60.

A powerful audio unified communication & collaboration solution for any organization, the UCM6300 Audio series provides a high-end unified communications solution packed with an ecosystem of mobility, security, voice and collaboration tools.

SKU: UCM6302A Category: Tag: Brand:

The Grandstream UCM6302A 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 end points. 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.

Grandstream UCM6302A Key Features

  • Supports 500 users and 75 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

Additional information

Specifications

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

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