Grandstream UCM6104 IP PBX System Dubai
The Grandstream UCM6104 Dubai is an advanced easy to manage IP PBX appliance for the SMB market with 2 FXS and 4 FXO Ports. Operating an improved version of Asterisk, the UCM6104 IP PBX Dubai contains advanced voice, data, video and mobility features without extra licensing or software fees.
The Grandstream UCM6104 Dubai is a powerful IP PBX appliance that supports up to 500 users, 3 conference bridges, 45 concurrent calls, and 25 conference members. This Grandstream UCM6100 Dubai Series features LAN/WAN Gigabit network port with combined PoE, up to 50 SIP trunk accounts with flexible call routing control, 4 FXO ports for POTS lines, 2 FXS ports for fax or analog phones, large 4GB onboard Flash memory and virtually unlimited looking for multi-site deployment.
The Grandstream UCM6104 Dubai series IP PBX appliance is built to bring enterprise-grade voice, data, video, and mobility features to small-to-medium companies within an easy-to-manage fashion. Integrating by including industry-leading features, the UCM6104 series offers easy and rapid setup and deployment while using web-browser interface – featuring auto-discovery of Grandstream endpoints and nil-Configuration provisioning.
The Grandstream UCM6104 Dubai series allows companies to join multiple communication technologies, such broad voice, interactive video, video calling, data tools, video surveillance, and facility access management onto one common network that that may be held and accessed remotely. The protected and reliable Grandstream UCM6104 Dubai series offers enterprise-grade features to SMBs in an unparalleled cost point cost without the certification costs, costs-per-feature, or recurring costs.
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Grandstream UCM6104 IP PBX Features:
- 1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing
- Integrated 4 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in case of power outage, and up to 50 SIP trunk accounts
- 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
- Supports up to 500 SIP endpoint registrations, up to 60 concurrent calls (up to 40 SRTP encrypted concurrent calls), and up to 32 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
- 2 FXS ports
- 4 FXO ports
- Dual Gigabit Ethernet ports
- USB, SD Peripheral Ports
- G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC, GSM; T.38 Voice and Fax Codecs
- H.264, H.263, H.263+ Video Codecs
- TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS Network Protocols
- Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom among other features