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Grandstream - UCM6202 IP PBX Series

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Stok: Tersedia

Harga Normal: Rp7.557.300,00

Special Price Rp6.345.000,00

Sinopsis

"New Series"
Mounting: Wall-mount & Desktop


  • UCM6202 supports up to 500 users/extension and 30 concurrent calls

  • Integrated 2 FXO ports for PSTN trunk and 2 FXS ports for analog telephone with lifeline capability

  • up to 50 SIP trunk accounts

  • up to 3 conference bridges with password-protected

  • up to 25 attendees per bridge simultaneous PSTN or IP participants

  • 2(two) 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 encyrption

  • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs

  • Auto Discovery and Zero Configuration of Grandstream SIP endpoints

Grandstream - UCM6202 IP PBX Series

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Product Overview

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.

Grandstream UCM6202 Key Features

  • Dual core 1GHz ARM CortexTM A9 and 400Mhz VINETICTM A8 processors, 1GB RAM and 4GB flash.
  • UCM6202 supports up to 500 users/extension and 30 concurrent calls
  • Integrated 2 FXO ports for PSTN trunk and 2 FXS ports for analog telephone with lifeline capability
  • up to 50 SIP trunk accounts
  • up to 3 conference bridges with password-protected
  • up to 25 attendees per bridge simultaneous PSTN or IP participants
  • 2(two) 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 encyrption
  • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
  • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
 

Grandstream UCM6202 - Technical Specifications:

FXS Ports for External Devices(analog telp, fax machine, etc)

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

FXO Ports for PSTN Line or any purpose

  • 2 ports fxo

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

  • 128x32 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

  • UCM6202: 226mm L x 155mm W x 34.5mm H

Weight

  • UCM6202: Unit weight 0.51kg, Package weight 0.94kg

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

  • -Registered SIP devices: supports up to  500 registered SIP devices/users
  • -Concurrent SIP calls: Up to 30 call or 66% performance if calls are SRTP encrypted

Conference Bridges

  • Up to 3(three) password-protected conference bridges allowing up to 25  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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