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VOIP PBX Products

Voice over IP PBX products offered by J. Darin Thomas Technology LLC.

Digium Switchvox 80 Appliance, 12 Concurrent Calls, One (1) Telephony Card Slot

Digium Switchvox 80 Appliance, 12 Concurrent Calls, One (1) Telephony Card Slot


  • Concurrent Users - up to 30
  • Concurrent Calls - up to 12
  • Telephony Card Slots - One (1)
  • PCI-Express Telephony Cards - 1TE133F, 1AEX(4,8)ELF, 1HB8-0000BLF
  • Redundant Hard Drive - No
  • Power Supply - Single 240W
  • Installation Hardware - 1U case, rack/wallmount brackets included
  • Warranty - One (1), Three (3), or Five (5) year warranty options
Switchvox 80 is ideal for small businesses with less than 30 users that need a shelf or desktop solution and a full-featured server-class PBX. Each Switchvox Unified Communications (UC) system supports VoIP and traditional calling while also incorporating many key features to enable your business communications.

Digium is the creator, sponsor, and innovative force behind Asterisk, the world's most popular open source telephony software. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. J. Darin Thomas Technology LLC provides full support services on all Digium products with our expertly trained engineers

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 


IP PBX Appliance

•Innovative IP Voice & Video


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


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 


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) 


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



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



NAT Router

Peripheral Ports

Voice-over-Packet Capabilities

Voice and Fax Codecs

Network Protocols


PSTN Line FXO Ports

Network Interfaces

Media Encryption


Customizable Auto Attendant

Call Features


Universal Power Supply

LCD Display

DTMF Methods

Provisioning Protocol & 


Call Center

Maximum Call Capacity

Conference Bridges


Voice/Video Capabilities

Signaling & Control

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



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


Switchvox unified communications on premise PBX products offered by J. Darin Thomas Technology LLC.

Custom implementations of Asterisk, the open source software pbx, offered by J. Darin Thomas Technology LLC.

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.

Jacksonville, Florida

Destin, Florida