Grandstream/PBX/8Line/3000 Users/3X Gigabit

Price:

5,950.00 QR


Panatron/PBX/24 Port
Panatron/PBX/24 Port
2,850.00 QR
2,850.00 QR

Grandstream/PBX/8Line/3000 Users/3X Gigabit

https://tamyeez.odoo.com/web/image/product.template/9710/image_1920?unique=82f4fbf
(0 review)

5,950.00 QR 5950.0 QAR 5,950.00 QR

5,950.00 QR

Not Available For Sale


This combination does not exist.


Share :
100% original guarantee
Return within 30days
Free delivery on all orders
Grandstream UCM6308 IP PBX.
The UCM6308 is an easy to manage device that does not require licensing fees and can support up to 6000 users and up to 400 concurrent calls. It features advanced technology including three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support for a NAT router. This enterprise-grade IP PBX can be integrated with third-party CRMs and PMS allowing for custom integrations. 

The Grandstream UCM6300 Series of IP PBX's have been designed with business of all sizes in mind, creating a unified communications solution that is robust and scalable. Moreover, the UCM6300 Series can be managed through the Grandstream Device Management System (GDMS) allowing for cloud-based setup, easy management and monitoring.

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.

Grandstream UCM6308 – Technical Specifications

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

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

  • 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 the 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, 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/ 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, 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

Grandstream UCM6308 – Technical Specifications

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

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

  • 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 the 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, 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/ 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, 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