Grandstream UCM6200 series IP PBX

by Grandstream

(KES 43,000 exc VAT) KES 49,880
  • UCM6202 and UCM6204 support up to 500 users and 50/75 concurrent calls, UCM6208 supports up to 800 users and 100 concurrent calls
  • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
  • Integrated 2/4/8 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
SKU9311

Reviews

This product does not have any reviews yet.

Add your review

Description

Grandstream UCM6200 series IP PBX 

Grandstream UCM6200 series IP PBX is 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.

 

UCM6200 series IP PBX 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.

 

UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees. Grandstream UCM6202 & 6204 IP-PBX appliances offer a complete solution to unify multiple communication technologies, like voice, video calling, video conferencing, video surveillance, data tools, mobility options and facility access management. The Grandstream UCM6202 has the following ports

Grandstream UCM6200 series IP PBX specifications

  • UCM6202 and UCM6204 support up to 500 users and 50/75 concurrent calls, UCM6208 supports up to 800 users and 100 concurrent calls
  • Auto Discovery and Zero Configuration of Grandstream SIP endpoints
  • Integrated 2/4/8 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
Analog Telephone FXS Ports 2 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports 4 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 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 Zero-Config (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, LADP
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 226mm L x 155mm W x 34.5mm H
Weight 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 Web UI: en,cn,es,fr,pt,de,ru,it,pl,cs Customizable IVR/voice prompts for en,cn,de,es,gr,fr,it,nl,pl,pt,ru,sv,tr,he,ar 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 45 or 66% performance if calls are SRTP encrypted
Conference Bridges Up to 3 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)