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Grandstream UCM6304 IP PBX

Grandstream

This series of IP PBX provides a platform that can integrate all business communications (including voice, video calls, video conferencing, video surveillance, web conferencing, data and analysis)

Grandstream UCM6304 Key Features

  • Supports  300 concurrent calls and about 2000 users
  • Ensure zero configuration of Grandstream SIP endpoints
  • Built-in conferencing platform and supports Wave app, desktop, and SIP endpoints
  • Wave for Android, iOS, Chrome and Firefox  browser permit interaction with all UCM6300 users & solutions
  • Availability of API for third-party integrations, including CRM and PMS platforms

$929.99

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$2,530.57 for 3 item(s)

Grandstream UCM6304 IP PBX

Grandstream UCM6304 IP PBX, companies can develop powerful and scalable unified communications and collaboration solutions. This series of IP PBX provides a platform that can integrate all business communications (including voice, video calls, video conferencing, video surveillance, web conferencing, data and analysis) into a centralized network. , Mobility, facility access, walkie-talkie, etc. The UCM6300 series can supports up to 3,000 users and it  includes an integrated web conferencing and video conferencing solution that enables employees to connect using desktop, mobile, GVC series devices, and IP phones.

The UCM6300 ecosystem consists of Wave applications for the Internet and mobile devices, which provide a hub for remote collaboration, and UCM RemoteConnect (NAT traversal service in the cloud) to provide secure remote connections. This UCM6300 series also provides cloud-based configuration and management through GDMS and API to be compatible with third-party platforms.  UCM6300 series provides high-quality unified communications and collaboration solutions, with a series of mobility, security, meeting and collaboration tools, and is a powerful platform for any enterprise.

Grandstream UCM6304 Key Features

  • Supports  300 concurrent calls and about 2000 users
  • Ensure zero configuration of Grandstream SIP endpoints
  • Built-in conferencing platform and supports Wave app, desktop, and SIP endpoints
  • Wave for Android, iOS, Chrome, and Firefox browsers permit interaction with all UCM6300 users & solutions
  • Availability of API for third-party integrations, including CRM and PMS platforms
  • Three-gigabit autosensing RJ45 network ports with built-in PoE + and compatible NAT router
  • Automated NAT firewall traversal service facilitates secure remote connections
  • Unique certificate and default random password to protect calls and accounts
  • It supports complete-Band Opus voice codec and H.264/H.263/ H.263+/H.265/VP8 video codec, jitter resilience up to 50% packet loss

Grandstream IP Phone Related Products

Grandstream GRP2601P VoIP Enterprise PhoneGrandstream GRP2601P VoIP Enterprise Phone
Grandstream GRP2601P 2-line Essential IP PhoneGrandstream GRP2601P VoIP Enterprise Phone

For Technical Support on this product, kindly complete our Contact US  Form, or If you require additional assistance with product installation, you can reach our support center at [email protected].

 

Grandstream UCM6304 – Technical Specifications

Brand

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

  • 4 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: 2490g
  • Package Weight: 3260g

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
  • 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/ work-load, in-queue announcement

Customizable Auto Attendant

  • Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Maximum Call Capacity

  • Users: 2000
  • Concurrent calls (G.711): 300
  • Max concurrent SRTP calls (G.711): 200

Maximum Attendees Of Conference Bridges

  • 4 Video Conference rooms and up to 40 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & G.711)
  • Voice Conference: Up to 200 parties (G.711)

Wave Mobile App

  • Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300

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