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Grandstream UCM6300A IP PBX Phone

Original price was: 15.050,00 EGP.Current price is: 12.980,00 EGP. 14% inc VAT

Sold by EG-tech

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Availability: In stock
SKU:UCM6300A
Grandstream, UCM6300A, IP PBX Phone,

Supports up to 250 users and up to 50 simultaneous calls
Provide zero configuration for Grandstream SIP endpoints.
Supports instant messaging (IM), voice conferencing and web conferencing from PCs, mobile devices, and SIP endpoints.
The free Wave app makes it easy to exchange voicemail and IM using desktop computers, the web and Android / iOS.
API is available for third-party integration, including CRM and PMS platforms.
Advanced security with secure boot, special certificate and default password to protect calls and accounts.
Three Gigabit RJ45 network ports with automatic sensor and PoE + and NAT router support
Automatic NAT firewall crossing facilitates remote connections.
Increases reliability with support for high frequency and high availability for two deployments.
Up to 50% support for Opus Full range audio codec, anti-vibration, packet loss.
Compatible with GDMS for cloud-based configuration, management and monitoring.

UCM6300 Audio Series

  • Supports up to 1500 users and up to 200 concurrent calls
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
  • Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
  • 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
  • Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
  • Supports Full-Band Opus voice 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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