Grandstream UCM6301 IP PBX System, Color : Black

₹ 45,000 / Piece

Brand NameGrandstream
ColorBlack
Voice-over-Packet CapabilitiesLEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation
Universal Power SupplyInput: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
Supply TypeExporter, Importer, Trader
Preferred Buyer Location All over the world

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ColorBlack
Brand NameGrandstream
Maximum Attendees of Conference Bridges2 Video Conference Rooms and Up to 12 Parties with 1080p, Assuming 4 Video Feeds + 1 Screen Sharing
Minimum Order QuantityAs per Customer Needs
MountingWall Mount & Desktop
Maximum Call CapacityUsers: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 50
Universal Power SupplyInput: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
Customizable Auto AttendantT Up to 5 Layers of Ivr (interactive Voice Response) in Multiple Languages
Voice and Fax CodecsOpus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.
Voice-over-Packet CapabilitiesLEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation
Telephony Operating SystemBased On Asterisk Version 16
Video CodecsH.264, H.263, H263+, H.265, VP8
APIFull API available for third-party platform and application integration

Features:

Supports up to 3000 users and up to 450 concurrent calls
Zero configuration provisioning of Grandstream SIP endpoints
Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
API available for third-party integrations, including CRM and PMS platforms
Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
Automated NAT firewall traversal service facilitates secure remote connections
Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/H.265/VP8 video codec, jitter resilience up to 50% packet loss
Compatible with GDMS for cloud setup, management and monitoring
Based on Asterisk* version 16 open source telephony operating system

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