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  »  Grandstream UCM6200 series IP PBX

Integrated 2/4/8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 50 SIP trunk accounts

Manufacturer: Grandstream Networks
Model: UCM6200 series
Datasheet

Optimized UC Features for SMBs

 


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.
 
Featuresbuy Grandstream UCM6200 series IP PBX from Network Options in Kenya

  • UCM6202 and UCM6204 support up to 500 users and 30/45 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 encyrption
  • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
     

Specification

Analog Telephone FXS Ports : 2 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports : 2 ports (UCM6202); 4 ports (UCM6204); 8 ports (UCM6208)
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
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