Unified Communications Platform

MIU (Magnet Interface Unit)

Magnetic interface cards are available for UCP1600/2120/4131 Series chassis; the MIU module provides 4/8 magnetic ports.

Product details

MIU Magnet Gateway, a powerful voice access device, provides IP network access for 4/8 channels of magnetic telephones. Using standard SIP protocols, it can interconnect with standard SIP softswitch systems to provide VoIP/MoIP solutions for users of magnetic telephones and others. The device is easy to deploy quickly, offers a high level of security and strong anti-jamming capability, and is widely used in fields such as military systems and other special command and dispatch systems.

The MIU panel supports 4/8 magnetic channel interfaces, a console interface, an operation status indicator, a power status indicator, channel status indicators, and an RST restart button. For its software interface, MIU uses the standard SIP protocol, is compatible with the most widely used IPPBX and SIP servers, and supports most VoIP operating platforms, such as Asterisk, Issabel, 3CX, FreeSwitch, BroadSoft, VOS, and so on.

Technical specifications:

Storage temperature range: -40~125℃

Operating temperature range: 0~50℃

Operating humidity range: 10%~90% non-condensing

Maximum power consumption: 6.5W

Dimensions: 12.4cm×17.3cm×2cm

Weight: 178g

Key features

Supports 4/8-port magnetic telephone interface

Supports WEB network management configuration

Supports various network protocols; the unified communications platform can be integrated with various types of interfaces

Connects to enterprise IP telephony systems and various unified communications systems to improve communication efficiency

Automatically detects and initiates a call to different magnetic telephone interfaces to establish a call quickly

Supports automatic hang-up for peer-to-peer calls and automatic hang-up for peer-to-peer calls.

The local magnetic telephone can ring to call the preset telephone; local hang-up requires pressing the control button.

The magnetic telephone interface uses transformer coupling mode, providing strong interference resistance.

Adapts to different network environments to ensure stable transmission.

Adaptive codec technology to ensure call quality

Rich logging features

Open API interface

Supports remote SSH operation

Supports configuration file backup and download

Supports DTMF detection for SIP v1 (RFC2543), v2 (RFC3261) protocols

Provides Ethernet electrical and optical interfaces, 10/100BASE-T full-duplex, half-duplex, adaptive, with bandwidth up to 100M

Supports DHCP, DNS/DDNS, NAT networking

Supports volume adjustment, gain adjustment, hold, call waiting, call transfer, and caller ID

Compatible with Asterisk, Elastix, 3CX, FreeSWITCH SIP Server, and VOS VoIP operating platforms

Specifications

See the manufacturer description and linked documentation for product information. No specification table is published on the captured product webpage.

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