The Yeastar TG200 is a VoIP gateway with 2 channels providing 4G network connectivity for soft switches, and IP-PBXs. It supports two-way communication: VoIP to 4G and 4G to VoIP.
- VoIP Gateway
- Connects 4G network to the VoIP network directly
- 2x 4G channels
- Carriers: AT&T
- Cost-effective backup when the landline goes down
- High compatibility with major IP-PBX and soft switch brands
- Transport: UDP, TCP, TLS, SRTP
- Voice Codecs: G.711 (alaw/ulaw), G.722, G.726, G.729A, WCDMA, ADPCM, Speex
- Echo Cancellation: ITU-T G.168 LEC
- LAN Ports: 1 10/100Base-T Ethernet
- Network Protocols: FTP, TFTP, HTTP, SSH
- Supports call transfer, call waiting, call status display, white list and black list, packet capture
Package Includes: Gateway, DC 12V, 1A power supply, warranty card, one short antenna.
Yeastar voIP Gateways
Cost-Effective & Feature-Rich
A Yeastar VoIP Gateway is an ideal addition to any SMB telephony system, bringing advanced VoIP technology & your phones and network together.
Yeastar's TA, TG, and TE Series VoIP Gateways offer up to 32 FXS ports, 16 FXO ports, 16 GSM/3G/4G channels, and two T1 ports.
Clear voice calls and carrier-grade reliability are guaranteed by a high-end TI chipset and processor. You'll get superb voice compressions.
Excellent interoperability with Asterisk, Lync Server (Skype for Business), FreePBX, Xorcom, 3CX platforms, and many more. Also certified with Elastix and Broadsoft.
The simple and intuitive web GUI allows for easy and straightforward configuration, saving you time.
A robust feature set that fulfills business needs & makes full use of analog phones, PSTN lines, digital lines, and cellular networks.
- 1 Stage/2 Stage Dial
- Call Back
- Call Duration Limitation
- Call Status Display
- Call Waiting
- 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
- White List and Black List
- Balance Alarm
- Call Detail Record (CDR)
- Call Progress Tone Generation
- Call Transfer
- Caller ID/CLIR
- Configure backup/restore
- Gain Adjustment
- IP Blacklist
- Packet Capture
- Real Open API Protocol (Based on Asterisk)
- Session Timer
- SIP Response Code Switch
- SIP Trunk: Support
- SMS Sending and Receiving
- Web based configuration