Innovative IP Voice & Video

The UCM6510 is an innovative IP PBX appliance for E1/T1/J1 networks that brings enterprise-grade Unified 

Communications and security protection to small-to-medium businesses (SMBs) 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. 

Feature Highlights 

1GHz quad-core Cortex A9 application processor, large memory 

(1GB DDR3 RAM, 32GB Flash), and dedicated high performance 

multi-core DSP array for advanced voice processing

1 integrated T1/E1/J1 interface, 2 PSTN trunk FXO ports, 2 analog 

telephone/Fax 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

Strong defense against malicious attacks (Fail2ban, Whitelist, 

Blacklist, alerts, etc.)

Supports up to 2000 SIP endpoint registrations, up to 200 

concurrent calls (up to 100 SRTP encrypted concurrent calls), 

and up to 64 simultaneous 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

Redundant power supply, advanced support for Hot Standby 

Clustering and High Availability (pending)

UCM6510 IP PBX Appliance 

UCM6510

IP PBX Appliance

•Innovative IP Voice & Video

UCM6510

Technical Specifications

Option 66/multicast SIP SUBSCRIBE/mDNS), eventlist between local and remote trunks

Multi-Language Support

English/Simplified Chinese/Traditional Chinese/Spanish/French/Portuguese/German/Russian/Italian/Polish/Czech 

for Web UI; Customizable IVR/voice prompts for English, Chinese, British English, German, Spanish, Greek, 

French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic

2 RJ11 ports (both with lifeline capability in case of power outage)

2 RJ11 ports (both with lifeline capability in case of power outage)

Dual Gigabit ports (switched or routed) with PoE;

A 3rd Gigabit port for Hot-Standby Clustering

Power 1/2, PoE, USB, SD, T1/E1/J1, FXS 1/2, FXO 1/2, LAN, WAN, Cluster Heartbeat

Yes, long press for factory reset and short press for reboot

Layer 3 QoS, Layer 2 QoS

SRTP, TLS, HTTPS, SSH

Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)

Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)

Unit Weight: 2.165 kg; Package Weight: 3.012 kg

Rack mount & Desktop 

Up to 5 layers of IVR (Interactive Voice Response)

Call park, call forward, call transfer, DND, DISA, ring group, pickup group, blacklist, paging/intercom etc.

FCC: Part 15 (CFR 47) Class B, Part 68

CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, TBR21, RoHS 

A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS 60950, AS/ACIF S002

ITU-T K.21 (Basic Level); UL 60950 (power adapter)

T1: TIA-968-B Section 5.2.4

E1: TBR12/TBR13, E1: AS/ACIF

Yes (user configurable) 

USB, SD

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

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

TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, 

SIP (RFC3261), STUN, SRTP, TLS, LDAP

Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A; 

128x32 dot matrix graphic LCD with DOWN and OK buttons

In Audio, RFC2833, and SIP INFO

TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP 

Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/

work-load, in-queue announcement

Up to 2000 registered SIP endpoints, up to 200 concurrent calls 

Up to 8 bridges, up to 64 simultaneous conference attendees

Analog Telephone FXS Ports

LED Indicators

Reset Switch

QoS

Mounting

NAT Router

Peripheral Ports

Voice-over-Packet Capabilities

Voice and Fax Codecs

Network Protocols

Physical

PSTN Line FXO Ports

Network Interfaces

Media Encryption

Environmental

Customizable Auto Attendant

Call Features

Compliance

Universal Power Supply

LCD Display

DTMF Methods

Provisioning Protocol & 

Plug-and-Play

Call Center

Maximum Call Capacity

Conference Bridges

Interfaces

Voice/Video Capabilities

Signaling & Control

Video Codecs H.264, H.263, H263+

Security

Physical

Additional Features

T1/E1/J1 Interface 1 RJ45 port

Polarity Reversal/Wink Yes, with enable/disable option upon call establishment and termination 

Caller ID Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT Japan 

Dimensions 440mm(L) x 185mm(W) x 44mm(H)

Disconnect Methods Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone

Digital Signaling PRI, SS7, MFC/R2

Advanced Defense Fail2ban, alert events, Whitelist, Blacklist, strong password based access control

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