Item Description

Quad Band 1x GSM SIM Gateway: Trunk to Asterisk iP PBX

GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cell phone roaming via the VoIP network.   

GoIP GSM Gateway is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks.    GoIP GSM Gateway is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GoIP GSM gateway also provides significant savings in usage (long distance or international), infrastructure and maintenance cost compared to conventional PSTN.




Benefits of GoIP

Extensive product compatibility with industry leading vendor

Cost-Savings on phone calls between mobiles or to PSTN

Easy to install – IP device with Web based management interface

Can be managed and monitored remotely over Internet - a great service to offer to customers by system integrators

GoIPs can be grouped together to establish GSM gateway cluster

Termination between GSM/VoIP

Schedule or on-demand SMS Broadcast messages to users

(Additional SMS server is required)

Key Features


Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)

Single Server Registrations

SIP peer-to-peer mode

SIP proxy mode

GSM module – 850MHz, 900 MHz, 1800 MHz, 1900MHz

Advanced jitter buffer

VLAN and QoS support

NAT Transversal

Call forward from GSM to VoIP and VoIP to GSM

Password or Trust list protection for dial in mode and dial out

Comprehensive dial Plan

Retransmit GSM Caller ID to VoIP terminal

Dynamic selection of codecs

Echo cancellation for Speakerphone

Comfort noise generation (CNG)

Voice activity detection (VAD)

Firmware upgrade from GUI






The HT-342 is designed as a compact, high performance, and low cost FXO Gateway.   The FXO detection is optimized to

avoid the hold up of the PSTN line when the other party is disconnected.   This has been one of the key issue in the design

of FXO gateway.   The incoming PSTN Caller ID is also  

transmitted to the VoIP user for more user friendly operation. The HT-342 is a full featured FXO gateway and is designed for

easy installation and configuration. It is an ideal solution for VoIP to PSTN termination in both SME and SOHO environment.

Key Features

Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)

Single or Multiple Server Registrations

Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer

Line Echo Cancellation

VLAN and QoS support

NAT Transversal and Router functions

Voice prompts, HTTP Web, Auto Provision support for configuration and updates

Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features

Two 10/100 Ethernet circuits connect to the LAN and an additional device

Four RJ-11 FXO ports for PSTN terminations (PSTN lines or PBX's extensions)

LEDs for Power, Ready, Status, WAN, PC, FXO ports  

Call forward from PSTN to VoIP and VoIP to PSTN

Dial in mode or dial out mode only

Dial Plan

Password protection for both PSTN dial in or dial out

Retransmit PSTN Caller ID to VoIP terminal

Enhanced Features

Dynamic selection of codec

Advanced jitter buffer

Automatic traversal of NAT and firewall

VLAN / Qos


Echo cancellation for Speakerphone

Comfort noise generation (CNG)

Voice activity detection (VAD)

Auto provisioning (requires auto provisioning server)

On line firmware upgrade

Multi-language support: English and Chinese

Supported Standards

ITU: H.323 V4, H.225, H.235, H.245, H.450


RFC 2327 – SDP

RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

RFC 2976 – SIP INFO Method

RFC 3261 – SIP

RFC 3264 – Offer/Answer model with SDP

RFC 3515 – SIP REFER Method

RFC 3842 – A Message Summary and Message Waiting Indicator

RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)

RFC 3891 – SIP “Replaces†Header

RFC 3892 – SIP Referred-By Mechanism

draft-ietf-sipping-cc-transfer-04 –   Session Initiation Protocol Call Control - Transfer

Codec: G.711 (A/µ law), G.729A/B, G.723.1

DTMF: RFC 2833, In-band DTMF, SIP INFO

Physical and Environmental

Operating temperature: 10°C to 40°C (50°F to 104°F)

Storage temperature: 0°C to 50°C (32°F to 122°F)

Weight:    1.9 kg (4.2 lb) (Including AC/DC Adapter)

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  • This item is covered by one year warranty directly from original factory
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