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.
Detail
| Models | TG100 | TG200 | TG400 | TG800 | TG1600 | |
| Number of Ports | 1 | 2 | 4 | 8 | 16 | |
| 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 suppot 4G LTE | Depending on the module type | ||||
| Protocol | SIP, IAX2 | |||||
| Antenna Splitter (4 in 1) | – | Support | ||||
| Transport | UDP, TCP, TLS, SRTP | |||||
| Voice Codec | G.711 (alaw/ulaw), G.722, G.726, G.729A, GSM, ADPCM, Speex | |||||
| DTMF Mode | RFC2833, SIP Info, In-band | |||||
| Echo Cancellation | ITU-T G.168 LEC | |||||
| Calling Type | Termination (VoIP to GSM/WCDMA), Origination (GSM/WCDMA to VoIP) | |||||
| Console Port | – | 1 | ||||
| Network Protocol | FTP, TFTP, HTTP, SSH | |||||
| LAN | 1 10/100 Mbps Ethernet Interface | 2 10/100 Mbps Ethernet Interfaces | ||||
| NAT Traversal | Static NAT, STUN | |||||
| Network | DHCP, DDNS, Firewall, OpenVPN, Static IP, QoS, Static Route, VLAN | |||||
| Operation Range | 0°C to 40°C, 32°F to 104°F | |||||
| Power Supply | DC 12V, 1A | AC 100-240V | ||||
| Storage Range | -20°C to 65°C, -4°F to 149°F | |||||
| Dimensions (L × W × H) (mm) | 110 x 70 x 24 | 213 x 160 x 44 | 340 x 210 x 44 | 440 x 250 x 44 | ||
| Humidity | 10-90% non-condensing | |||||
Key Features
• 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