Here is a properly structured, SEO-optimized product description for the Grandstream UCM6510 IP PBX with support for 2000 SIP extensions. This content adheres to your formatting and encoding guidelines and avoids all specified errors.

---

Grandstream UCM6510 IP PBX with Support for 2000 SIP Extensions

The Grandstream UCM6510 is an enterprise-grade IP PBX designed for high-volume voice, video, data, and mobility communications. It supports up to 2000 SIP extensions and 200 concurrent calls, making it ideal for large businesses, call centers, and institutions that require a reliable, scalable, and secure communication system. The UCM6510 integrates traditional telephony with advanced VoIP services, combining T1/E1/J1 digital trunk support with analog FXO/FXS interfaces. With zero license fees, robust security protocols, and easy deployment features, it delivers maximum flexibility and cost-efficiency for large-scale unified communications.

Product Features

High Capacity and Scalability

Supports up to 2000 SIP endpoint registrations and up to 200 simultaneous SIP calls, providing scalability for growing enterprises and multi-site operations.

Digital and Analog Trunk Integration

Includes one T1/E1/J1 digital trunk port, two FXO ports for analog PSTN lines, and two FXS ports with lifeline support, ensuring full compatibility with legacy telephony infrastructure.

Advanced Conference and Collaboration Tools

Supports up to 64-way video conferences and 200-way audio conferences, with built-in features such as call bridging, multi-language IVRs, voicemail-to-email, and call recording.

Secure Unified Communication Platform

Equipped with TLS, SRTP, HTTPS, SSH, and fail2ban intrusion prevention, offering enterprise-grade security to protect sensitive communication data.

Zero-Configuration Provisioning

Supports automatic discovery and configuration of Grandstream IP endpoints, simplifying system deployment and reducing IT workload.

Redundant Power and Failover Options

Includes dual network ports with failover and NAT functionality, plus internal call routing to maintain communication continuity in case of network failure.

Centralized Web-Based Management

Offers an intuitive web GUI for administration, system monitoring, and call reporting, including real-time call detail records, logs, and alerting functions.

License-Free Advanced Features

All system features are included without the need for additional licenses, subscriptions, or per-user fees, ensuring predictable budgeting and maximum ROI.

Product Specifications

• Supports up to 2000 SIP extensions
• Supports up to 200 concurrent SIP calls
• 1 × T1/E1/J1 digital trunk interface
• 2 × FXO ports for PSTN lines
• 2 × FXS ports with analog lifeline pass-through
• Dual Gigabit Ethernet ports with PoE+ support
• Built-in NAT router and firewall
• 1 × USB port for storage or peripheral connection
• 1 × SD card slot for backup and recording storage
• Integrated LDAP and XML phonebook support
• Voicemail and fax-to-email functionality
• Multi-level IVR (up to 5 levels) with multi-language support
• Call queues, ring groups, paging and intercom features
• Audio codec support: G.711 A/u-law, G.722, G.726, G.729A/B, iLBC, Opus
• Video codec support: H.264, H.263, H.263+
• Secure protocols: TLS, SRTP, HTTPS, SSH, fail2ban
• Onboard LCD status display with operational controls
• Internal power supply: 100–240 VAC, 50–60 Hz input
• Output: 12 VDC, 1.5 A
• Dimensions: 440 mm × 185 mm × 44 mm (1U rack mount)
• Operating temperature: 0°C to 45°C
• Humidity: 10% to 90% non-condensing
• Complies with FCC, CE, and RoHS standards

---

If you need a version formatted for integration with an e-commerce CMS (like WooCommerce or Shopify), or product meta descriptions for SEO purposes, I can provide those as well.