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Grandstream GS-UCM6102 VoIP PBX 2x FXS and 2x FXO

Grandstream GS-UCM6102 VoIP PBX 2x FXS and 2x FXO

GS-UCM6102 (Stock only sold to existing CT clientbase)

  • Supports up to 500 users, 50 SIP trunk accounts, up to 30 concurrent calls
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Strongest-possible security protection using SRTP, TLS and HTTPS encryption
  • Dual Gigabit network ports with integrated PoE
  • Supports up to a 5-level IVR (Interactive Voice Response)
  • Built-in call recording server; recordings accessed via web user interface
  • Supports call queue for efficient call volume management
  • Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
  • Multi-language auto-attendant to efficiently handle incoming calls
  • Integrated LDAP and XML phonebooks, flexible dial plan
  • Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
  • Supports voicemail and fax forwarding to email
  • 1 Year Carry In Warranty

R 4089.00 incl. VAT on Prepaid

More Information:

The UCM6100 Series is an innovative IP PBX appliance designed to bring enterprise-grade Unified Communications and Security Protection features to small-to-medium businesses (SMBs) in an easy-to-manage fashion. Powered by an advanced hardware platform and revolutionary software functionalities, the UCM6100 Series offers a breakthrough turnkey solution for converged voice, video, data, fax, security surveillance, and mobility applications out of the box without any extra license fees or recurring costs.

1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing Integrated 2/4/8/16 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in case of power outage, and up to 50 SIP trunk accounts.

Gigabit network port(s) with integrated PoE, USB, SD card; integrated NAT router with advanced QoS support (UCM6102 only) Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based caller ID/call progress tone and smart automated impendance matching for various countries.

Supports up to 500 SIP endpoint registrations, up to 60 concurrent calls (up to 40 SRTP encrypted concurrent calls), and up to 32 conference attendees Flexible dial plan, call routing, site peering, call recording, central control panel for endpoints, integrated NTP server, and integrated LDAP contact directory Automated detection and provisioning of IP phones, video phones, ATAs, gateways, SIP cameras, and other endpoints for easy deployment Strongest-possible security protection using SRTP, TLS, and HTTPS with hardware encryption accelerator.

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