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UCM series VoIP PBX from Grandstream.

Grandstream UCM6100 series IP PBX Appliance

Grandstream’s UCM6100 series is an open source, licensing-free SMB IP PBX appliance for delivering secure and reliable voice, video, data and mobility apps. Powered by an advanced hardware platform based on Asterisk, the UCM6100 series helps smaller organizations to affordably use VoIP to increase productivity, provide better customer service, unify communications on a single platform and save money on communications costs. Open source systems ensure compliancy to SIP-based protocols meaning the IP PBX can work with a range of desktop SIP endpoints (IP phones, video cameras, etc.) as well as popular service providers, SIP trunk providers and other SIP hardware. Open source solutions also provide flexibility to easily develop and customize applications to fit business integration, interoperability and communications needs.

$99.00 $90.00
Quantity:

Grandstream UCM6510 - Two (2) Port IP PBX Appliance with 200 Concurrent Calls

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

$1,999.00 $1,909.00
Quantity:

Grandstream UCM6116 - 16 Port IP PBX Appliance with 60 Concurrent Calls

Grandstream UCM6116 - 16 Port IP PBX Appliance with 60 Concurrent Calls

 

  • PSTN Line FXO Ports - 16
  • Analog Telephone FXS Ports - Two (2), both with lifeline capability
  • Concurrent Calls - Up to 60
  • Conference Bridges - Up to 6, with 32 simultaneous PSTN or IP participants
  • SIP Trunk Accounts - Up to 50
  • Network Interfaces - Dual 10M/100M/1000M RJ45 Ethernet ports
  • Peripheral Ports - USB, SD (up to 32GB)
  • Quality of Service - Layer 3 QoS
A standalone Asterisk Open Source IP PBX appliance. Voice-over-Packet Capabilities include LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, and Modem detection and auto-switch to G.711. Multi-language support. Advanced security features built in firewall with SRTP/TLS encryption, 802.1X network security and HTTPS Web UI. No licensing or recurring fees. All hardware, software features, and functions included. Free lifetime firmware updates. Two (2) year manufacturer warranty. *Device can be powered via PoE.
$1,799.00 $1,709.99
Quantity:

Grandstream UCM6108 - Eight (8) Port IP PBX Appliance with 60 Concurrent Calls 

Grandstream UCM6108 - Eight (8) Port IP PBX Appliance with 60 Concurrent Calls

 

  • PSTN Line FXO Ports - Eight (8)
  • Analog Telephone FXS Ports - Two (2), both with lifeline capability
  • Concurrent Calls - Up to 60
  • Conference Bridges - Up to 6, with 32 simultaneous PSTN or IP participants
  • SIP Trunk Accounts - Up to 50
  • Network Interfaces - Dual 10M/100M/1000M RJ45 Ethernet ports
  • Peripheral Ports - USB, SD (up to 32GB)
  • Quality of Service - Layer 3 QoS
A standalone Asterisk Open Source IP PBX appliance. Voice-over-Packet Capabilities include LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, and Modem detection and auto-switch to G.711. Multi-language support. Advanced security features built in firewall with SRTP/TLS encryption, 802.1X network security and HTTPS Web UI. No licensing or recurring fees. All hardware, software features, and functions included. Free lifetime firmware updates. Two (2) year manufacturer warranty. Device can be powered via PoE.
$949.00 $909.99
Quantity:

Grandstream UCM6104 - Four (4) Port IP PBX Appliance with 45 Concurrent Calls

Grandstream UCM6104 - Four (4) Port IP PBX Appliance with 45 Concurrent Calls

 

  • PSTN Line FXO Ports - Four (4)
  • Analog Telephone FXS Ports - Two (2), both with lifeline capability
  • Concurrent Calls - Up to 45
  • Conference Bridges - Up to 3, with 25 simultaneous PSTN or IP participants
  • SIP Trunk Accounts - Up to 50
  • Network Interfaces - Dual 10M/100M/1000M RJ45 Ethernet ports
  • Peripheral Ports - USB, SD (up to 32GB)
  • Quality of Service - Layer 3 QoS
A standalone Asterisk Open Source IP PBX appliance. Voice-over-Packet Capabilities include LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, and Modem detection and auto-switch to G.711. Multi-language support. Advanced security features built in firewall with SRTP/TLS encryption, 802.1X network security and HTTPS Web UI. No licensing or recurring fees. All hardware, software features, and functions included. Free lifetime firmware updates. Two (2) year manufacturer warranty. Device can be powered via PoE.
$499.00 $489.99
Quantity:

Grandstream UCM6102 - Two (2) Port IP PBX Appliance with 30 Concurrent Calls

Grandstream UCM6102 - Two (2) Port IP PBX Appliance with 30 Concurrent Calls

 

  • PSTN Line FXO Ports - Two (2)
  • Analog Telephone FXS Ports - Two (2), both with lifeline capability
  • Concurrent Calls - Up to 30
  • Conference Bridges - Up to 3, with 25 simultaneous PSTN or IP participants
  • SIP Trunk Accounts - Up to 50
  • Network Interfaces - Dual 10M/100M/1000M RJ45 Ethernet ports
  • Peripheral Ports - USB, SD (up to 32GB)
  • Quality of Service - Layer 3 QoS
  • Integrated NAT Router with advanced QoS support - UCM6102 only
A standalone Asterisk Open Source IP PBX appliance. Voice-over-Packet Capabilities include LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, and Modem detection and auto-switch to G.711. Multi-language support. Advanced security features built in firewall with SRTP/TLS encryption, 802.1X network security and HTTPS Web UI. No licensing or recurring fees. All hardware, software features, and functions included. Free lifetime firmware updates. Two (2) year manufacturer warranty. Device can be powered via PoE.
$399.00 $389.99
Quantity: