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Grandstream UCM6116 IP PBX

Up to 60 concurrent conversations, 32 conference participants, 16 FXO ports, and 2 FXS ports are supported by the UCM 6116 VoIP SIP PBX. The UCM6116 IP PBX system is a full-featured, easy-to-manage IP PBX Appliance that brings enterprise-grade features to small and medium-sized businesses (SMBs) without the need for licence fees, feature prices, or recurring costs.

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Description

The Grandstream UCM6116 IP PBX appliance is a revolutionary IP PBX appliance that brings enterprise-grade Unified Communications and Security Protection to small-to-medium enterprises (SMBs) in an easy-to-manage manner. The UCM6116, driven by an innovative hardware architecture and novel software functionalities, provides a breakthrough turnkey solution for integrated voice, video, data, fax, security surveillance, and mobility applications right out of the box.

Grandstream’s UCM6116 is the most recent model in the UCM 6100 series. Its features are similar to those of other phone systems in the same series, but with its exceptional features, efficient capabilities, and updated applications, it is capable of leading the industry.The UCM6116 IP PBX is a full-featured, easy-to-manage IP PBX Appliance that gives small and medium-sized businesses (SMBs) enterprise-grade services without the need for licence fees, feature prices, or recurring costs.

Features

  • 1GHz ARM Cortex A8 application processor, plenty of memory (512MB DDR RAM, 4GB NAND Flash), and a dedicated high-performance multi-core DSP array for advanced voice processing.
  • 16 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in the event of a power outage, and up to 50 SIP trunk accounts are all integrated.
  • Gigabit network port(s) with PoE, USB, and SD card support; integrated NAT router with advanced QoS support (UCM6102 only).
  • For various countries, hardware DSP-based 128ms tail-length carrier-grade line echo cancellation (LEC), hardware-based caller ID/call progress tone, and smart automated impedance matching are available.
  • Up to 500 SIP endpoint registrations, 60 concurrent calls (40 SRTP encrypted concurrent calls), and 32 conference attendees are supported.
  • Call routing, site peering, call recording, centralized control panel for endpoints, integrated NTP server, and integrated LDAP contact directory are all available.
  • IP phones, video phones, ATAs, gateways, SIP cameras, and other endpoints are automatically detected and provisioned for easy deployment.
  • SRTP, TLS, and HTTPS are used in conjunction with a hardware encryption accelerator to provide the highest level of security possible.

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Amit singh – Acmatel Pvt Ltd

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