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العملاء قامو بزيارة
Grandstream UCM6301 IP PBX UCM6301
- The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions.
- This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more.
- The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones.
- It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution.
- The UCM6300 ecosystem consists of the Wave app for web and mobile, which provides a hub for collaborating remotely, and UCM Remote Connect, a cloud NAT traversal service for ensuring secure remote connections.
- The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-party platforms.
- By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300 series provides a powerful platform for any organization.
- 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
|النوع||هاتف أي بي|
|Maximum Call Capacity||
Users- 500 Concurrent calls (G.711)
|Maximum Attendees of Conference Bridges||
2 Video Conference rooms and up to 12 parties with 1080p; assuming 4 video feeds + 1 screen sharing
|Customizable Auto Attendant||
Up to 5 layers of IVR (Interactive Voice Response) in multiple languages
Wall mount & Desktop
|Universal Power Supply||
Input 100 ~ 240VAC; 50/60Hz; Output DC 12V; 1.5A
|Telephony Operating System||
Based on Asterisk version 16
Full API available for third-party platform and application integration
H.264; H.263; H263+; H.265; VP8
|Voice and Fax Codecs||
Opus; 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
LEC with NLP Packetized Voice Protocol Unit; 128ms-tail-length carrier grade Line Echo Cancellation
|Minimum Order Quantity||
|الضمان||1 year warranty|