TG VoIP GSM Gateway

TG Series VoIP GSM Gateways

Yeastar TG Series VoIP GSM Gateways connect GSM or WCDMA or 4G LTE to VoIP networks to provide two-way communication: GSM/3G/4G to VoIP and VoIP to GSM/3G/4G. This allows you to connect most IP-based telephone systems including Yeastar IP Phone Systems, and softswitches to a GSM or 3G WCDMA or 4G LTE; which can provide a sophisticated fallback solution when landlines go down, or be used to increase call traffic capacity by providing additional dial-tone.

► GSM Voice Calls
► VoLTE for HD Calls
► Maximum Cost Reduction
► Top Quality and Reliability
► Mobile Connectivity for SMB

Easy to use

Simple and intuitive Web-based configuration saves you loads of time

Mobile Trunkings

Add GSM/WCDMA trunkings for business making high number of calls to mobile networks

Bulk SMS Service

Great tool for enterprises to manage customer relations and introduce special offers with a low cost


Interoperable with a broad list of softswitch, PBX, and IP-PBX like Elastix and Lync Server
Number of Ports124816
GSM Frequency**850/900/1800/1900 MHz
WCDMA Frequency**850/1900 MHz, 850/2100 MHz, 900/2100 MHz
4G Data
4G LTE Band**Don’t support 4G LTEDepending on the module type. Check the supported band and operators.Don’t support 4G LTE
ProtocolSIP, IAX2
Antenna Splitter (4 in 1)Support
TransportUDP, TCP, TLS, SRTP
Voice CodecG.711 (alaw/ulaw), G.722, G.726, G.729A, GSM, ADPCM, Speex
DTMF ModeRFC2833, SIP Info, In-band
Echo CancellationITU-T G.168 LEC
Calling TypeTermination (VoIP to GSM/WCDMA), Origination (GSM/WCDMA to VoIP)
Console Port1
Network ProtocolFTP, TFTP, HTTP, SSH
LAN1 10/100 Mbps Ethernet Interface2 10/100 Mbps Ethernet Interfaces
NAT TraversalStatic NAT, STUN
NetworkDHCP, DDNS, Firewall, OpenVPN, Static IP, QoS, Static Route, VLAN
Operation Range0°C to 40°C, 32°F to 104°F
Power SupplyDC 12V, 1AAC 100-240V
Storage Range-20°C to 65°C, -4°F to 149°F
Dimensions (L × W × H) (mm)110 x 70 x 24213 x 160 x 44340 x 210 x 44440 x 250 x 44
Humidity10-90% non-condensing
  • 1 Stage/2 Stage Dial
  • Call Back
  • Call Duration Limitation
  • Call Status Display
  • Carrier Selection: Auto/Manual
  • Firmware upgrade by HTTP/TFTP
  • GSM/CDMA/UMTS Ports Group Manage
  • Incoming /Outgoing Routing rules
  • Network Attack Alert
  • Open API for SMS and USSD
  • PIN Modify
  • Send Bulk SMS
  • SIP Peer Mode: Support
  • SIP server for IP phones: Support
  • SMS Center
  • System Logs
  • VoIP Trunk Group
  • Black List
  • Balance Alarm
  • Call Detail Record (CDR)
  • Call Progress Tone Generation
  • Call Transfer
  • Caller ID/CLIR
  • Configure backup/restore
  • Gain Adjustment
  • Hotline
  • IP Blacklist
  • NTP
  • Packet Capture
  • Real Open API Protocol (Based on Asterisk)
  • Session Timer
  • SIP Response Code Switch
  • SIP Trunk: Support
  • SMS Sending and Receiving
  • USSD
  • Web based configuration