VOIP Products
VOIP Products
Voice over IP products offered by J. Darin Thomas Technology LLC.
- Details
- Parent Category: VOIP PBX Products
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
- Details
- Parent Category: VOIP PBX Products
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
- Details
- Parent Category: VOIP PBX Products
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
- Details
- Parent Category: VOIP PBX Products
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
Subcategories
VOIP PBX Products
Voice over IP PBX products offered by J. Darin Thomas Technology LLC.
Switchvox from Digium
Switchvox unified communications on premise PBX products offered by J. Darin Thomas Technology LLC.
Custom Asterisk Designs
Custom implementations of Asterisk, the open source software pbx, offered by J. Darin Thomas Technology LLC.
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.
VOIP Telephones
Voice over IP Phones offered by J. Darin Thomas Technology LLC.
Grandstream
Grandstream Networks is a manufacturer of IP voice/video telephony and video surveillance solutions. Grandstream serves the SMB with innovative products that lower communication costs, increase security protection and enhance productivity. Their open standard SIP-based products offer broad industry interoperability and price-performance competitiveness. NETXUSA is an official Grandstream distributor providing full support for their entire product line.
Digium Telephones
This family of high-definition Digium IP phones is designed for the greatest interoperability with Asterisk or Switchvox.
These phones fully leverage the power of Asterisk, the world’s most widely adopted open source communications software, and Switchvox, Digium’s award-winning Unified Communications (UC) system. With Digium technology on both the server and the phone, customers will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system – all at an extremely competitive price.
VOIP Accessories
VOIP Accessories offered by J. Darin Thomas Technology LLC.
Headphones
Telephone Headphones offered by J. Darin Thomas LLC.