Grandstream UCM6308 IP PBX System

1,017,630.00

  • 3000 users
  • 450 concurrent calls
  • 3 Gigabit Ports
  • PoE Support
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The Grandstream UCM6308 IP PBX System is a highly advanced and sophisticated communication device designed to provide a comprehensive and high-quality solution for unified communications. Equipped with a quad-core 1.3GHz ARM Cortex A53 processor, 2GB RAM, and 128GB Flash, this system delivers high-performance operations without compromising speed or efficiency. Additionally, it supports up to 3000 users, making it perfect for large enterprises. Key features of the Grandstream UCM6308 IP PBX System include 8 PSTN Line FXO Ports, 2 analog telephone FXS Ports, and up to a 5-level Interactive Voice Response (IVR).

For increased security measures, this system also incorporates TLS and SRTP security encryption technology to protect calls and accounts. An added benefit of the UCM6308 is its support for video conferencing, thanks to its built-in video recording capabilities. This feature allows businesses to conduct face-to-face meetings regardless of geographical location, providing an efficient and effective solution for remote collaboration.

The Grandstream UCM6308 IP PBX System comes with zero configuration provisioning, ensuring easy installation and maintenance. With its auto-discovery and zero-configuration capabilities, it eliminates the complexity usually associated with establishing VoIP networks.

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Feature Description
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 2x USB 3.0
1x 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
Customizable IVR/voice prompts English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Netherlands
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
Customizable 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, provides a centralized interface to provision, manage, monitor and troubleshoot Grandstream products