Grandstream UCM6302A
- Supports up to 500 users and up to 75 concurrent calls
- Zero configuration provisioning of Grandstream 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
- 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
- Description
- Specifications
Description
Audio VoIP PBX Solutions in Pakistan – Grandstream UCM6302A:
The Grandstream UCM6302A is an Audio VoIP PBX voice system that supports up to 500 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App that allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints. It supports UCM Remote Connect cloud service for remote users to offer a best-in-class hybrid platform that combines the control of an on-premise IP PBX with the remote access and system manageability of a cloud solution. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, instant messaging, voice conferencing, and collaboration tools, the UCM6300 Audio series provides a powerful business communication platform for any organization.
Specification
|
Analog Telephone FXS Ports |
2 RJ11 ports |
|
Peripheral Ports |
1*USB 2.0, 1*USB 3.0, 1*SD card interface |
|
LCD Display |
320×240 color LCD with touch screen for Shortcut Keys and Scroll Bar |
|
Reset Switch |
Yes, long press for factory reset and short press for reboot |
| QoS | Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS |
| Telephony Operating System | Based on Asterisk version 16 |
| DTMF Methods | In-band audio, RFC4733, and SIP INFO |
| Universal Power Supply |
● Input: 100 ~ 240VAC, 50/60Hz
● Output: DC 12V, 1.5A |
| Dimension (W x D x H) | 270mm(L) x 175mm(W) x 36mm(H) |
| Media Encryption |
SRTP, TLS, HTTPS, SSH, 802.1X
|
