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Grandstream UCM6308 IP PBX

  • Up to 3000 users
  • Up to 450 concurrent calls
  • 8 Video Conference rooms
Overview

The Grandstream UCM6308 allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralised network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 series 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 IP PBX with the remote access of a cloud solution.

Grandstream UCM6308 Key Features

  • Supports up to 3000 users and up to 450 concurrent calls
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
  • Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
  • 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
  • Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/H.265/VP8 video 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
Features

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

  • 8 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

  • 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

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

  • 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: 2550g
  • Package Weight: 3320g

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
  • 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: 3000
  • Concurrent calls (G.711): 450
  • Max concurrent SRTP calls (G.711): 300

Maximum Attendees of Conference Bridges

  • 8 Video Conference rooms and up to 60 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & G.711)
  • Voice Conference: Up to 300 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
Whats in the Box
  • Grandstream UCM6308 IP PBX
  • User Guide
Shipping

Next Day Delivery
For UK orders received before 12pm Monday to Friday, we operate a next business day delivery service, subject to availability. We use next day services from the leading delivery companies.

Specific Time Delivery
If you need an item urgently, or delivered at a specific time, please contact us.

Delivery Methods
We use several leading delivery companies. The majority of our deliveries are made by
DPD, including most international orders.

Shipping prices are calculated during the checkout process.
We deliver to almost every country. For international delivery services, we use DHL, a high quality, fast delivery service. Prices are provided based on the weight of the order; this is calculated when you reach the checkout.

For full shipping information please visit our shipping page.

VoIP Packages

Hosted Business VoIP Provider Unlimited Calls from £5/month
If you’re looking for a reliable and competitive VoIP provider for your small business, we can offer you one of the UK’s most trusted services. No need to compromise on quality, either. The capabilities of our VoIP PBX system have been developed over a decade, with our roadmap for the future incorporating even more improvements. Your business VoIP is in safe hands with us.

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SIP Trunking

With SIP trunking, your business can operate its own on-site IP PBX system. Xinix has been offering our own SIP trunk solutions for over 10 years.

Business SIP trunking is a service that enables your company to operate a selection of telecommunications systems over your IP network. It offers a number of benefits – from cheaper international call costs to improved business flexibility. Incorporating a Disaster Recovery strategy to give access to an increased quantity of numbers, business SIP trunking from Xinix World can enhance the way your business stays connected.

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£2,203.32 Excl. VAT

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  • In Stock
  • Order within 11 hours for delivery on Mon, 27th May