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Manufacturers > Manufacturers GrandStream > UCM6102 innovative IP PBX appl
Part No : GS-UCM6102

UCM6102 innovative IP PBX appl

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Our Price
:
$ 290.99
List Price
:
$ 340.99
In Stock
:
0
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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 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- 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
 
Features
  • 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 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
  • 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