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Grandstream UCM6308A Audio Series IP PBX Scalable Business Communication & Collaboration Platform

Grandstream

Datasheet_UCM6300_Audio_Series_English

UCM6304-6308-6308A_QIG-1

  • Supports up to 1,500 users and 200 simultaneous calls
  • Easy provisioning of Grandstream SIP endpoints
  • Built-in IM, Audio & Web Conferencing: Accessible via desktop, mobile, or SIP devices
  • Communicate from anywhere on Android, iOS, or Web
  • API support for CRM, PMS, and more
  • Secure boot, unique device certificates, and random default passwords
  • With PoE+ and integrated NAT router functionality Remote
  • Automated NAT firewall traversal for secure remote access High
  • Hot Standby and local dual deployment support Superior Audio

$807.57

Grandstream UCM6308A Audio Series IP PBX Scalable Business Communication & Collaboration Platform

The Grandstream UCM6308A Audio Series IP PBX is a next-generation unified communications solution designed to meet the evolving needs of modern businesses. This powerful and scalable IP PBX system seamlessly integrates voice, video, instant messaging (IM), conferencing, mobility, facility access, and more—offering a comprehensive communication platform for enterprises of all sizes.

Built on the reliable Asterisk version 16* open-source telephony system, the UCM6300 Audio Series supports up to 1,500 users and 200 concurrent calls, delivering enterprise-grade performance without licensing fees. It features built-in instant messaging, voice/web conferencing, and integrates effortlessly with Grandstream’s free Wave App, enabling teams to stay connected across desktops, smartphones, SIP phones, and other compatible devices.

The UCM6308A is more than just an IP PBX—it’s a complete unified communication platform that enhances team collaboration, boosts productivity, and strengthens data security. Whether you’re running a small business or managing a large-scale enterprise, this system adapts to your growth and changing communication needs.

Key Features of Grandstream UCM6308A

  • Supports up to 1,500 users and 200 simultaneous calls
  • Easy provisioning of Grandstream SIP endpoints
  • 8 FXO ports and 2 FXS ports for analog line integration
  • Built-in IM, Audio & Web Conferencing: Accessible via desktop, mobile, or SIP devices
  • Communicate from anywhere on Android, iOS, or Web
  • API support for CRM, PMS, and more
  • Secure boot, unique device certificates, and random default passwords
  • Supports PoE, AES encryption, and advanced call routing
  • Automated NAT firewall traversal for secure remote access High
  • Hot Standby and local dual deployment support Superior Audio
  • Full-band Opus codec with up to 50% packet loss resilience Cloud
  • Compatible with Grandstream Device Management System (GDMS)
SKU: UCM6308A-1 Categories: , , , , , , , , , , , , Tags: , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , , Brand:
Brands

Color

Network connection type

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

  • 8 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

  • Power 1/2, FXS, FXO, LAN, WAN, Heartbeat

LCD Display

  • 128×32 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

  • 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

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

Dimensions

  • 485mm(L) x 187.2mm(W) x 46.2mm(H)

Weight

  • Unit Weight: 2538g
  • Package Weight: 3463g

Temperature & Humidity

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

Mounting

  • Rack 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: 1500
  • Concurrent calls (G.711): 200
  • Max concurrent SRTP calls (G.711): 150

Maximum Attendees of Conference Bridges

  • 9 meeting rooms and up to 150 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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