Grandstream UCM6208 Series IP PBX Appliance

750.00

This IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video conferencing, video surveillance, data tools, mobility options and facility access management onto one common network.

Available on backorder

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SKU: UCM6208 Category: Tags: , , ,

Description

Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video conferencing, video surveillance, data tools, mobility options and facility access management onto one common network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.

The same industry-leading platform that we designed for the UCM6100 series, which continues to be enhanced month by month with more and more features and functionalities, will be extended to the UCM6200 series. Looking for a more powerful upgrade? We have redesigned our 8 FXO port model, the UCM6208, to offer the ability to support more users and more concurrent calls, as it supports up to 800 users and up to 100 concurrent calls.

  • UCM6208 supports up to 800 users and 100 concurrent calls
  • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
  • Integrated 8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts
  • Gigabit network ports with Integrates PoE, USB, SD card
  • Supports up to a 5-level IVR (Interactive Voice Response)
  • Built-in call recordings server; recordings accessible via web user interface
  • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
  • Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
  • Strongest possible security protection using SRTP, TLS and HTTPS encryption
  • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs

Additional information

Analog Telephone FXS Ports

2 ports (both with lifeline capability in case of power outage)

PSTN Line FXO Ports

8 ports

Network Interfaces

Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009)

NAT Router

Yes (supports router mode and switch mode)

Peripheral Ports

USB, SD

LED Indicators

Power/Ready, Network, PSTN Line, USB, SD

LCD Display

128×32 graphic LCD with DOWN & OK button

Reset Switch

Yes

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

Voice and Fax Codecs

G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC, GSM, AAL2-G.726-32,
ADPCM; T.38

Video Codecs

H.264, H.263, H263+

QoS

Layer 3 QoS, Layer 2 QoS

DTMF Methods

In Audio, RFC2833, and SIP INFO

Provisioning Protocol & Plug-and-Play

TFTP/HTTP/HTTPS, 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,
SIP (RFC3261), STUN, SRTP, TLS, LDAP

Disconnect Methods

Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone

Media Encryption

SRTP, TLS, HTTPS, SSH

Universal Power Supply

Output: 12VDC, 1.5A; Input: 100 ~ 240VAC, 50 ~ 60Hz

Dimensions

440mm L x 185mm W x 44mm H

Weight

Unit weight 2.23kg, Package weight 3.09kg

Environmental

Operating: 32 ~ 104ºF / 0 ~ 40ºC, 10 ~ 90% (non-condensing); Storage: 14 ~ 140ºF / -10 ~ 60ºC

Mounting

Wall mount & Desktop

Multi-Language Support

English/Simplified Chinese/Traditional Chinese/Spanish/French/Portuguese/German/Russian/
Italian/Polish/Czech for Web UI; Customizable IVR/voice prompts for English, Chinese, British
English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish,
Turkish, Hebrew, Arabic; Customizable language pack to support any other languages

Caller ID

Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT

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/busy level, in-queue announcement

Customizable Auto Attendant

Up to 5 layers of IVR (Interactive Voice Response)

Maximum Call Capacity

– 800 registered SIP devices/users
– Up to 100 concurrent SIP calls

Conference Bridges

Up to 6 password-protected conference bridges allowing up to 32 simultaneous PSTN or IP participants

Call Features

Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom etc.

Compliance

FCC: Part 15 (CFR 47) Class B, Part 68
CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, TBR21, RoHS
A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, AS/ACIF S002
ITU-T K.21 (Basic Level); UL 60950 (power adapter)

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