Description

Grandstream UCM6510 IP PBX Dubai

The Grandstream UCM6510 series allows businesses to unify multiple communication technologies, such as voice, video, surveillance, data tools, and facilities access management into one common platform that can be managed and accessed remotely. With features such as customisable call-routing, multi-level IVRs, call queues, auto-attendant, call detail records, multi-site peering, SIP video support, voicemail/fax forwarding to email and more, the Grandstream UCM6510 Dubai delivers complete unified communications.

Grandstream UCM6510 IP PBX Features

High Performance & Stability

The Grandstream UCM 6510 IP PBX Dubai comes with a distributed architecture that configures an integrated media resource card. It also not only features a highly responsive concurrent call processing, but also a smart auto load balancing function which results in an excellent performance, thus giving you a maximum communication experience.

Multi-Language User Interface

The Grandstream UCM6510 IP PBX Dubai is designed to support many different languages including 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, and Arabic.

Supports Power over Ethernet

The Grandstream UCM6510 IP PBX Dubai has equipped Gigabit Ethernet ports which are integrated with Power over Ethernet to allow for convenient installation and a variety of remote features including provisioning, status monitoring and handset firmware upgrades.

Cloud Management

Grandstream provides a reliable and secure cloud service to deliver connectivity to its users at cheaper maintenance costs while enhancing operation and maintenance efficiency. The cloud service on the internet enables customers to gain direct access to Grandstream devices available in private networks such as IP-PBXs, audio gateways, and IP phones, therefore allowing simple remote maintenance and management of Grandstream devices. This kind of cloud service is specifically designed to meet the needs of large-scale installation, configuration, and maintenance and operation. Auto-provisioning and configuration backup, online update, real-time monitoring and alert enable users to achieve efficient operation and maintenance, thus maximizing productivity in organizations.

Flexible Deployment, Easy Operation & Simple Maintenance

The Grandstream UCM 6510 IP PBX system Dubai comes with a centralized single-site deployment and multi-site distributed network. Its centralized equipment management ensures a highly efficient operation and maintenance. The highly efficient Grandstream UCM 6510 IP PBX system Dubai uses a graphical configuration interface to accord you with an easy user experience even without any technical expertise.

Supports 2 FXO Ports and 2 FXS Ports

Foreign Exchange Subscribers is the port that actually delivers the analog line to the subscriber. It is simply the “plug in the wall” that delivers a dial tone, battery current and ring voltage. This is the jack or interface to the phone system to which FXO devices can be connected to. The Grandstream UCM6510 IP PBX Dubai has been equipped with 2 FXS ports to maximize communication while using your telephone to make calls, thus maximizing work productivity. It also comes with 2 FXO ports and allows for the making of up to 200 concurrent VoIP calls. It supports up to 2000 SIP devices or users, proving to be a highly efficient device in busy offices.

Automated – Provisioning – TFTP/HTTP/HTTPS

The Grandstream UCM 6510 IP PBX Dubai has the ability to deploy an information technology or telecommunications service by using embedded pre-deﬁned procedures that are carried out electronically without requiring any human intervention, thus making it very reliable and convenient.

Rich QoS Capabilities

Quality of Service (QoS) is a feature of routers and switches which prioritizes traffic so that more important traffic can pass first, resulting in an overall performance improvement for critical network traffic. This feature is very useful in the Grandstream UCM 6510 IP PBX Dubai whenever there are high volumes of traffic, therefore eliminating any delays while working.

Excellent Audio Quality

The high performing Grandstream UCM 6510 IP PBX Dubai offers users distraction-free communication courtesy of the inbuilt industry-leading Acoustic Echo Canceling Technology, Adaptive Jitter Buffer, Voice Activity Detection, and Comfort Noise Generation that deliver superior High Definition sound quality devoid of extraneous noises. This makes it possible to enjoy fluent phone conversations without any distractions.

SIP – Based Connections

You can easily connect your Grandstream UCM 6510 IP PBX Dubai to your IP PBX System or direct to a VoIP service provider in order to enjoy seamless conference calls. The Grandstream UCM 6304 IP PBX Dubai is based on SIP and works perfectly with most of the SIP platforms such as Cisco CallManager/CUCM, Broadsoft, Microsoft Skype for Business (Lync), – Huawei IMS, and Asterisk/Elastix that currently exist in the market.

Firmware Upgradeable

You can easily update the operating system on your Grandstream UCM 6510 IP PBX Dubai to improve its functionality and enhance user experience. Upgrading firmware also fixes any existing bugs and protects you from any kind of software malfunctions and security threats. To update your gadget’s firmware, type your gateway’s IP address into your web browser and enter your login information. Then locate the Firmware or Update section and download the latest firmware update on your Grandstream UCM 6510 IP PBX system’s manufacturer’s website. Finally, upload the update and reboot the telephony system.

Call Features

Call Transfer: This advanced feature of the Grandstream UCM 6510 IP PBX system Dubai allows the user to relocate an inbound call to another phone or messaging system by using a dedicated call transfer button, or software that has been configured for use on the Gateway.
Call Waiting: You can hear another incoming call when you are already on an active phone call (beep). With the Grandstream UCM6510 IP PBX, you can also turn off call waiting so that incoming calls are directly sent to voicemail during moments you are active on another phone call.
Call Holding: You can easily place an active phone call on hold in order to make or pick another incoming call using the premium Grandstream UCM6510 IP PBX.

Grandstream UCM6510 Key Features

Supports up to 2000 SIP endpoint registrations, up to 200 concurrent calls and up to 64 conference attendees
1GHz quad-core Cortex A9 processor
1GB DDR3 Ram, 32GB Flash
1 Integrated T1/E1/J1 interface, 2PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with lifeline capability
Gigabit network ports with Integrates PoE, USB, SD card, integrated NAT router
Comprehensive security protection using SRTP, TLS and HTTPS with hardware encryption accelerator
Quickly set up and provision Grandstream endpoints using the Auto-Discovery and Zero Config feature within the product’s web user interface.

Grandstream UCM6510 IP PBX Specifications

Interfaces

Analog Telephone FXS Ports 2 RJ11 ports (both with lifeline capability in case of a power outage)
PSTN Line FXO Ports 2 RJ11 ports (both with lifeline capability in case of a power outage)
T1/E1/J1 Interface 1 RJ45 port
Network Interfaces Dual Gigabit ports (switched or routed) with PoE; A 3rd Gigabit port for Hot-Standby Clustering
NAT Router Yes (user configurable)
Peripheral Ports USB, SD
LED Indicators Power 1/2, PoE, USB, SD, T1/E1/J1, FXS 1/2, FXO 1/2, LAN, WAN, Cluster Heartbeat
LCD Display 128×32 dot matrix graphic LCD with DOWN and OK buttons
Reset Switch Yes, long press for factory reset and short press for reboot

Voice/Video Capabilities

Voice-over-Packet Capabilities 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
Voice and Fax Codecs 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
Video Codecs H.264, H.263, H263+
QoS Layer 3 QoS, Layer 2 QoS

Signaling & Control

In Audio, RFC2833, and SIP INFO In Audio, RFC2833, and SIP INFO
PRI, SS7, MFC/R2, RBS (pending) PRI, SS7, MFC/R2, RBS (pending)
Provisioning Protocol & Plug-and-Play TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66/multicast SIP SUBSCRIBE/mDNS), eventlist between local and remote trunks
Network Protocols TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261),
STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending)
Disconnect Methods Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect, Busy Tone

Security

Media Encryption SRTP, TLS, HTTPS, SSH
Advanced Defense Fail2ban, alert events, Whitelist, Blacklist, strong password-based access control

Physical

Universal Power Supply Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A;
Physical Unit Weight: 2.165 kg; Package Weight: 3.012 kg
Dimensions 440mm(L) x 185mm(W) x 44mm(H)
Environmental Operating: 32 – 113ºF / 0 ~ 45ºC, Humidity 10 – 90% (non-condensing) Storage: 14 – 140ºF / -10 ~ 60ºC, Humidity 10 – 90% (non-condensing)
Mounting Rackmount & Desktop

Additional Features

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
Caller ID Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT Japan (pending)
Polarity Reversal/Wink Yes, with enable/disable option upon call establishment and termination
Call Center Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/work-load, in-queue announcement
Customizable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity Up to 2000 registered SIP endpoints, up to 200 concurrent calls
Conference Bridges Up to 8 bridges, up to 64 simultaneous conference attendees