• 2 SIP lines
  • Dual 10/100 Mbps RJ45 Ethernet ports
  • One-touch paging
  • G.722 wideband audio decoding

The Becke GP100 is a compact SIP paging microphone designed for fixed communication positions that do not require a numeric keypad or touchscreen. A single programmable speed-dial key provides direct access to a configured SIP destination, allowing an operator to start a paging or voice communication task without entering a number manually.

With two SIP lines, PoE, dual Ethernet ports, wideband G.722 audio and full-duplex voice communication, the GP100 is suited to desks where the communication path is predefined. It can be used at reception counters, security desks, plant offices, gatehouses and other positions where operators repeatedly contact the same paging or intercom destination.

GP100 one-touch SIP paging microphone with gooseneck microphone for desktop voice paging

One-Touch Operation

The GP100 removes the numeric keypad and display found on more interactive paging terminals. Instead, the operating workflow is based around a dedicated physical key and a predefined SIP destination.

This approach is useful where the operator does not need to search a directory, select different zones from a touchscreen or dial multiple extensions throughout the day. Once the required destination has been configured, the terminal can provide a direct path to that voice service.

Speed Dial

The physical key supports one-touch speed dialing. Depending on the surrounding SIP platform and configuration, the programmed destination may represent an extension or another supported voice service.

Keeping the destination predefined also reduces the number of operating steps at positions where the same communication path is used repeatedly.

Status Indication

A green status indicator provides local information for SIP and network status. Since the GP100 has no screen, this indicator gives the operator a simple visual reference for basic terminal connectivity.

SIP Voice Communication

Although its controls are deliberately simple, the GP100 includes a full SIP voice feature set behind the front panel. It supports two SIP lines and can be used for paging-oriented calls as well as two-way intercom communication.

Two SIP Lines

Two SIP lines allow the terminal to be registered according to the voice architecture of the project. This provides more flexibility than a single-purpose analog paging microphone and allows the GP100 to operate as a network voice endpoint within a SIP-based communication system.

Voice Intercom

The terminal supports two-way voice intercom and configurable automatic answering. Automatic answering can be set individually for supported lines, which is useful when the GP100 is required to receive direct communication without additional local operation.

Three-Way Call

Local three-way conference calling is also supported. This allows an operator to bring an additional SIP participant into a supported voice session when a simple two-party conversation is not sufficient.

GP100 SIP paging microphone with two SIP lines dual Ethernet one-touch paging and G.722 wideband audio

Audio Performance

A paging microphone is only useful if spoken instructions remain intelligible at the receiving side. GP100 therefore supports both narrowband and wideband SIP audio codecs, together with voice-processing functions intended to improve two-way communication.

Wideband Audio

Supported codecs include G.711 A/U, G.729A/B, iLBC, G.723, G.726-32K and G.722. G.722 provides a wideband option for compatible SIP endpoints and servers where higher voice bandwidth is required.

Echo Control

Full-duplex communication is supported with acoustic echo cancellation. The specification provides echo-delay cancellation of up to 96 ms in hands-free operation, helping reduce acoustic return during two-way conversations.

Voice Processing

Additional audio processing includes voice activity detection, comfort noise generation, background noise processing and packet-loss compensation. These functions work at the terminal level to help maintain usable voice communication under different network and acoustic conditions.

Network Deployment

GP100 is designed to fit into an Ethernet-based voice network without requiring a separate analog paging cable from every desk position back to the central system.

Dual Ethernet

Two 10/100 Mbps RJ45 interfaces are provided: one network port and one PC port. Network bridge mode allows another Ethernet device to share the local network connection where the project design permits it.

This can be useful at reception desks, offices and duty positions where the available network outlet must serve both the paging microphone and another desktop device.

PoE Power

The WAN interface supports IEEE 802.3af Class 3 Power over Ethernet. With a compatible PoE switch, network connectivity and power can therefore be supplied through the same Ethernet cable.

A DC 12V / 1A power input is also provided for installations that use a local power adapter instead of PoE.

Web Management

Remote web-based volume adjustment is supported, allowing selected terminal settings to be managed over the network rather than requiring an installer to visit every paging position for routine adjustment.

Network time synchronization is also supported for system environments where consistent terminal time is required.

External Integration

GP100 can participate in more than basic SIP calling. Audio output and relay I/O provide additional connection options for projects that need to integrate the desktop terminal with external equipment.

Audio Output

A 3.5 mm three-pole audio output provides an adjustable signal of up to 1.2 Vpp. It can be connected to compatible active speakers or other suitable external audio equipment when additional local audio output is required.

Relay I/O

The terminal provides a one-channel short-circuit input/output interface with relay. The relay is specified for a maximum of DC 24V / 1A or AC 120V / 1A and uses a pluggable 3.5 mm Phoenix terminal.

This interface can be incorporated into project-specific control or event linkage. The actual linked function should be defined by the surrounding system design rather than assuming the GP100 directly controls a particular device.

Deployment Roles

GP100 is most appropriate when the location has a clearly defined communication role. It is not intended to replace a touchscreen paging console that needs frequent destination selection.

Reception Desks

At a reception or visitor-service desk, the GP100 can provide a dedicated button for contacting a predefined paging or communication service without requiring the receptionist to operate a separate dialing interface.

Security Posts

Gatehouses, security desks and duty rooms can use the terminal where staff regularly need to contact the same control point or initiate a predefined voice communication path.

Industrial Offices

Plant offices, warehouse coordination desks and workshop support positions can use the GP100 as a fixed SIP voice endpoint for routine announcements or communication with other compatible SIP terminals.

Service Counters

Hospitals, transport facilities and public-service counters can deploy the terminal where operators require direct voice access but do not need a numeric keypad, directory or touchscreen at the desk.

GP100 one-touch SIP paging microphone used at a fixed reception and announcement desk

Technical Specifications

Product Model GP100 One-Touch SIP Paging Microphone
Version Non-keypad desktop version
SIP Lines 2
Network Interface 2 × RJ45, including network and PC interfaces with bridge-mode support
PoE IEEE 802.3af Class 3 on WAN interface
DC Power DC 12V / 1A, 5.5 mm power jack
Status Indicator 1 × green indicator for SIP and network status
Operation One-touch speed dial through dedicated physical key
Voice Intercom Supported
Auto Answer Supported and configurable by SIP line
Conference Local three-way conference calling
Voicemail Supported with compatible server
Voice Codecs G.711A/U, G.729A/B, iLBC, G.723, G.726-32K, G.722
Echo Cancellation Full-duplex AEC with up to 96 ms echo-delay cancellation in hands-free mode
Audio Processing VAD, CNG, background noise processing and packet-loss compensation
DTMF In-band, RFC2833 and SIP INFO
Audio Output 1 × 3.5 mm audio output, maximum 1.2 Vpp, adjustable volume
Relay Interface 1-channel short-circuit input/output with relay
Relay Rating Maximum DC 24V / 1A or AC 120V / 1A
Microphone Integrated gooseneck microphone
Scheduled Tasks Supports scheduled local periodic tasks
Remote Management Supports remote web-based volume adjustment
Time Synchronization Network time synchronization
Housing Aluminum alloy
Installation Desktop
Operating Temperature -30°C to +60°C
Relative Humidity 10% to 95%

FAQ

Does GP100 have a numeric keypad?

No. This GP100 version does not include a numeric keypad or touchscreen. It uses a dedicated physical key for one-touch speed dialing to a configured SIP destination.

Is GP100 a paging microphone or a SIP phone?

GP100 is best positioned as a one-touch SIP paging microphone and desktop voice terminal. It uses SIP calling technology but is designed around a fixed microphone and predefined voice workflow rather than general telephone dialing.

Can GP100 receive calls?

Yes. The terminal supports voice intercom and configurable automatic answering. Auto-answer behavior can be configured according to the SIP line and system requirements.

Does GP100 support PoE?

Yes. The WAN interface supports IEEE 802.3af Class 3 PoE. A DC 12V / 1A input is also available as an alternative power method.

Can GP100 connect to an external speaker?

Yes. A 3.5 mm adjustable audio output is provided for connection to compatible active speakers or other suitable external audio equipment.

Specifications

                Item                Specification
                Product Model                GP100 Non-Keypad SIP Microphone
                Network Interface                2×RJ45 interfaces (1×Network, 1×PC, support network bridge mode); WAN port supports PoE power supply (IEEE 802.3af, class 3)
                Power Interface                1×DC power interface, input: DC12V / 1A; interface specification: 5.5mm DC power jack (Φ5.5×2.1×10mm)
                Status Indicator                1×green indicator (for SIP status and network status)
                Audio Output Interface                1×3.5mm standard 3-segment audio interface, max output 1.2Vpp (volume adjustable), for connecting active speakers
                Dry Contact Interface                1-channel short-circuit input/output interface (with relay); relay specification: MAX DC24V/1A, AC120V/1A; interface type: pluggable 3.5mm phoenix terminal
                Housing Material                Aluminum alloy housing
                Installation Method                Desktop placement
                Color                Black + White
                Operating Temperature                -30℃ ~ 60℃
                Storage Temperature                -30℃ ~ 60℃
                Relative Humidity                10%~95%
                Voice Codec                G.711A/U, G.729A/B, iLBC, G.723, G.726-32K, G.722
                Echo Cancellation                Full-duplex communication with AEC (Acoustic Echo Cancellation) algorithm, max 96ms echo delay cancellation in hands-free mode
                Audio Enhancement Technologies                VAD (Voice Activity Detection), CNG (Comfort Noise Generation), BNE (Background Noise Elimination), PLC (Packet Loss Compensation)
                DTMF Transmission Mode                In-band, Out-of-band (RFC2833/SIP INFO)
                SIP Lines                2 lines
                Microphone Configuration                Equipped with gooseneck microphone for excellent pickup performance
                Time Management                Support scheduled local periodic tasks
                Remote Management                Support remote web-based volume adjustment
                Time Synchronization                Support network time synchronization
                Voice Intercom                Supported
                Speed Dial                Supported (one-touch activation via speed dial key)
                Auto Answer                Supported (configurable for individual lines)
                Voicemail                Supported (requires server support)
                Local 3-Way Conference Call                Supported


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