UCM6510 - User Manual
WELCOME
Thank you for purchasing Grandstream UCM6510 IP PBX appliance. The UCM6510 is an innovative IP PBX appliance for E1/T1/J1 networks that brings
enterprise-grade unified communications and security protection to enterprises, small-to-medium businesses (SMBs), retail environments and residential
settings in an easy-to-manage fashion. Powered by an advanced hardware platform and revolutionary software functionalities, the UCM6510 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.
Caution
Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User
Manual, could void your manufacturer warranty.
Warning
Please do not use a different power adaptor with the UCM6510 as it may cause damage to the product and void the manufacturer warranty.
Info
This document is subject to change without notice. The latest electronic version of this user manual is available for download here:
http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of
Grandstream Networks, Inc. is not permitted.</p>
PRODUCT OVERVIEW
Feature Highlights
1 integrated E1/T1/J1 interface, 2 FXO ports, and 2 FXS ports with lifeline capability in case of power outage.
Hardware DSP-based carrier-grade line echo cancellation (LEC) with 128ms-tail-length, hardware-based caller ID/call progress tone, and automatic
impedance matching for various countries.
Gigabit network ports with the LAN port supporting PoE, USB 2.0 port, SD card slot, and an integrated NAT router with advanced QoS support.
Several protective measures against malicious attacks: Fail2Ban, whitelisting, blacklisting, alerts, etc.
Data and data-voice communication via E1/T1/J1 with SS7/PRI.
Supports up to 2000 SIP endpoint registrations, 200 concurrent calls (132 SRTP encrypted concurrent calls), and 64 conference participants.
Offers flexible dial plans, call routing, site peering, and call recordings (manual/automatic for SIP calls).
Functions as a central control panel for endpoints, integrated NTP server, and integrated LDAP contact directory.
Automated detection and provisioning of supported IP phones, video phones, ATAs, gateways, SIP cameras, and others for simple and quick deployment
Secure encryption with SRTP, TLS, and HTTPS with hardware encryption accelerator.