Description
GRANDSTREAM UCM6510 IP PBX System: The Definitive Enterprise Communication Hub for the UAE
In the dynamic and competitive business environment of the United Arab Emirates, particularly in major hubs like Dubai and Abu Dhabi, large enterprises require communication systems that are not only powerful and feature-rich but also scalable, reliable, and cost-effective. The GRANDSTREAM UCM6510 IP PBX System emerges as the definitive solution, meticulously engineered to meet these demanding requirements. Designed specifically for organizations leveraging E1, T1, or J1 digital telephony circuits, the Grandstream UCM6510 provides a robust bridge between traditional telecommunications infrastructure and modern Voice over IP (VoIP) technology, delivering a comprehensive suite of unified communication (UC) features without the encumbrance of per-user licensing fees.
At its core, the Grandstream UCM6510 is built for performance and scale. It supports an impressive capacity of up to 2000 registered SIP users or endpoints and can effortlessly manage up to 200 concurrent calls. This substantial capacity makes it an ideal choice for large corporate offices, bustling call centers, expansive hospitality venues, educational institutions, and government bodies across the UAE. Whether your organization is experiencing rapid growth or operates with consistently high call volumes, the UCM6510 provides the headroom needed to ensure seamless communication flow without bottlenecks or performance degradation. The foundation of this capability lies in its powerful hardware architecture: a 1GHz quad-core Cortex A9 processor, a generous 1GB of DDR3 RAM, and 32GB of NAND Flash memory. This combination ensures swift call processing, smooth multitasking between various UC features like conferencing and recording, and a highly responsive management interface, guaranteeing a superior user experience and system stability.
Seamless Integration with Digital Telephony Infrastructure
A key differentiator for the Grandstream UCM6510 is its native support for digital telephony trunks through its integrated T1/E1/J1 interface. This allows businesses in Dubai and Abu Dhabi to directly connect the IP PBX to PRI, SS7, or MFC R2 circuits provided by local telecom operators like Etisalat or du. This integration offers several advantages over purely SIP trunk-based solutions, particularly for established enterprises. It often provides higher call quality guarantees, enhanced reliability, and allows organizations to leverage existing telecom contracts and infrastructure investments. The UCM6510 acts as a powerful gateway, translating calls between the digital network and the IP network, enabling the use of modern SIP endpoints and UC features while maintaining connectivity through traditional, highly reliable digital lines. Furthermore, the inclusion of 2 FXO ports provides vital PSTN lifeline capability, ensuring that basic call functionality can be maintained even in the event of an internet outage or primary trunk failure – a critical consideration for business continuity.
A Comprehensive Suite of Unified Communication Features
The Grandstream UCM6510 transcends basic call handling, offering a rich ecosystem of unified communication tools designed to enhance productivity and collaboration:
- Advanced IVR and Auto Attendant: Create sophisticated call routing menus with up to 5 levels of Interactive Voice Response. This allows businesses to automate call handling, direct callers efficiently to the right department or individual, provide information automatically, and project a highly professional image 24/7.
- Integrated Call Center Capabilities: Manage inbound call flows effectively with features like Automatic Call Distribution (ACD) based on agent skills, availability, or workload. Implement call queues with customizable announcements and music-on-hold to manage caller wait times efficiently. These features are invaluable for customer service departments, technical support centers, and sales teams operating within the UAE.
- Built-in Call Recording: Record calls effortlessly for quality assurance, compliance purposes (adhering to relevant UAE regulations), dispute resolution, or staff training. The Grandstream UCM6510 provides an integrated recording server, eliminating the need for separate hardware or software.
- Robust Conferencing: Host multi-party audio conferences with ease. The UCM6510 supports up to 8 conference bridges, accommodating a total of 64 simultaneous participants. This facilitates collaboration between teams located in different offices within the UAE or internationally.
- SIP Video Support: Embrace visual communication by integrating SIP video endpoints. The UCM6510 supports H.264, H.263, and H.263+ video codecs, enabling face-to-face meetings and video calls directly through the PBX system.
- Mobility Features: Extend office communication to mobile devices using compatible softphone applications, ensuring employees stay connected whether they are in the office, working remotely from other Emirates, or traveling.
- Zero-Configuration Provisioning: Simplify the deployment of Grandstream SIP endpoints (IP phones, video phones, ATAs). The Grandstream UCM6510 can automatically detect and configure compatible devices on the network, significantly reducing setup time and IT workload.
Uncompromising Security and Reliability
Security is paramount in enterprise communications. The Grandstream UCM6510 incorporates multiple layers of protection to safeguard conversations and system integrity. Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS) encrypt voice and signaling traffic, preventing eavesdropping. HTTPS ensures secure web-based management access. Advanced defense mechanisms like Fail2ban automatically block IP addresses exhibiting malicious behavior, while configurable whitelists and blacklists provide granular access control. Strong password policies further enhance security. For mission-critical environments demanding maximum uptime, the UCM6510 supports a high-availability configuration where a secondary, redundant UCM6510 unit can take over automatically in case the primary unit fails, ensuring minimal disruption to business operations.
Flexible Networking and Management
Integrating the UCM6510 into existing network infrastructure is straightforward. It features dual Gigabit Ethernet ports with integrated Power over Ethernet Plus (PoE+), capable of powering connected IP phones or other PoE devices directly. These ports can operate in either switch mode (for simple integration) or router mode (providing NAT routing capabilities). Management is handled through an intuitive, multilingual web-based graphical user interface (GUI), allowing administrators to easily configure extensions, trunks, call routes, IVRs, and other system settings. Comprehensive Call Detail Records (CDR) provide valuable insights into call patterns and usage.
The Smart Investment for UAE Enterprises
For large organizations in Dubai, Abu Dhabi, and across the UAE seeking a powerful, scalable, and feature-rich IP PBX system designed for digital trunking, the GRANDSTREAM UCM6510 presents an exceptional value proposition. It delivers enterprise-grade performance and a comprehensive UC feature set comparable to significantly more expensive systems, but without the ongoing burden of licensing fees. Its robust hardware, extensive capacity, seamless integration capabilities, strong security posture, and ease of management make it the ideal foundation for a modern, reliable, and future-proof business communication strategy. Choosing the UCM6510 is an investment in enhanced productivity, improved collaboration, and operational efficiency for any large-scale enterprise operating in the demanding UAE market.
GRANDSTREAM UCM6510 IP PBX System – Specifications
- Analog Telephone FXS Ports: 2 ports (both with lifeline capability in case of power outage)
- PSTN Line FXO Ports: 2 ports
- T1/E1/J1 Interface: 1 port
- Network Interfaces: Dual Gigabit RJ45 ports with integrated PoE Plus (IEEE 802.3at-2009)
- NAT Router: Yes (supports router mode and switch mode)
- Peripheral Ports: USB, SD
- LED Indicators: Power 1/2, PoE, USB, SD, T1/E1/J1, FXS 1/2, FXO 1/2, LAN, WAN
- 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-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: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:264, H.263, H263+
- QoS: Layer 3 QoS, Layer 2 QoS
- DTMF Methods: In Audio, RFC2833, and SIP INFO
- Digital Signaling: T1: PRI, SS7, MFC/R2; E1: PRI, SS7, MFC/R2 (pending)
- Provisioning Protocol & Plug-and-Play: TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP endpoints via Zero-Config (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
- 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
- Media Encryption: SRTP, TLS, HTTPS, SSH
- Advanced Defense: Fail2ban, alert events, Whitelist, Blacklist, strong password based access control
- Universal Power Supply: Output: DC +12V, 1.5A; Input: 100 ~ 240VAC, 50/60Hz
- 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: Rack mount & Desktop
- 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; Customizable language pack to support any other languages
- Caller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT
- 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
- Call Features: Call park, call forward, call transfer, DND, DISA, ring group, pickup group, blacklist, paging/intercom etc.
- Compliance: FCC: Part 15 (CFR 47) Class B, Part 68; CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, TBR21, RoHS; RCM: AS/NZS CISPR 22, 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: TBR4/TBR12/TBR13, E1: AS/ACIF