Yeastar Neogate TG200 – 2 Port GSM VoIP Gateway
Yeaster Neogate TG200 GSM VoIP gateway Connect GSM or WCDMA or 4G LTE to VoIP networks to provide two-way communication. This allows you to connect most IP-based telephone systems including Yeastar, Grandstream, Yealink and other IP Phone Systems
Yeastar Neogate TG200 Features Review
- Number of GSM channels (Max): 2
- Network type: 850/900/1800/1900MHz
- Protocol: SIP (RFC3261), IAX2
- Transport Protocol: UDP,TCP,TLS,SRTP
- DTMF: RFC2833, SIP INFO, In-band
- Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM, ADPCM, Speex
- Echo Cancellation: ITU-T G.168 LEC
- LAN: 1 (10/100 Mbps)
- Network: Static IP, DHCP Client, Firewall, VLAN, DDNS, QoS, OpenVPN
- NAT Traversal: Static NAT, STUN
- Size: 110x70x24mm
- Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)
Use Cases
Bulk SMS Service.
TG Bulk SMS feature is a great tool in implementing SMS campaigns. With TG, sending bulk SMS is only clicks away. Enter the desired phone numbers, the SMS content in the TG Web GUI, there you go! The Bulk SMS feature is easy to use, fast, and reliable. TG also provides API to connect an external SMS client.
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TG as a SIP Register
TG adds GSM trunks to transform fixed-to-mobile calls to mobile-to-mobile calls. It could significantly reduce telecommunication expenses. Meanwhile, TG brings value-added features such as SMS, callback, and call forwarding, etc. The GSM trunking is also a low-cost alternative to landlines in instances where fixed-line communication is not available.
Yester TG 200 GSM Gateway Bebefits:
- GSM Voice Calls
- VoLTE for HD Calls
- Mobile Connectivity for SMB
- Top Quality and Reliability
- Maximum Cost Reduction
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
- Block List
- Balance Alarm
- Call Detail Record (CDR)
- Call Progress Tone Generation
- Call Transfer
- Caller ID/CLIR
- Configure backup/restore
- Gain Adjustment
- Hotline
- IP Blocklist
- 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
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