Yeastar TG200 VoIP GSM Gateway

The Yeastar TG200 VoIP GSM Gateway is a dual-channel GSM/CDMA/UMTS gateway that connects mobile networks to SIP-based phone systems. It helps businesses cut telephony costs while providing reliable voice calls, SMS, and data services. Additionally, it offers backup call routing, ensuring your office communication stays stable even during network issues. This VoIP GSM gateway is ideal for small and medium businesses, offices, and PBX systems that need efficient and secure mobile connectivity.

Key Benefits

• Two GSM channels (quad‑band 850/900/1800/1900 MHz) for making and receiving voice/SMS over mobile networks.

• Enables VoIP‑to‑mobile and mobile‑to‑VoIP communication, ideal for least-cost routing (LCR) and mobile trunking.

• Acts as a cost-effective failover when fixed‑line networks fail.

• Easy setup and integration with web-based management interface and compatibility with major IP‑PBX and softswitch platforms such as Yeastar, Asterisk, FreePBX, 3CX, Elastix.

Yeastar TG200 VoIP Technical Specs & Features

• Protocols: SIP (RFC3261), IAX2 for flexible VoIP deployment.

• Audio codecs: G.711a/u, G.722, G.726, G.729A, GSM, ADPCM, Speex; echo cancellation via ITU‑T G.168.

• Network support: Static IP / DHCP, NAT traversal (STUN), firewall, DDNS, VLAN, QoS, OpenVPN support.

• Transport and DTMF: UDP, TCP, TLS, SRTP; DTMF via RFC2833, SIP INFO, in-band.

• Hardware: One 10/100 Mbps LAN port, external antenna connectors (2), AC 100–240 V power input. Package includes power cord, network cable, antennas, mounting hardware.

Use Cases

• Least Cost Routing — Direct traffic over GSM lines to reduce expensive international or mobile-to-mobile calls.

• Emergency Backup — Automatically switch to mobile trunking during ISP or PSTN outages.

• SMS Broadcasting — Send bulk SMS or USSD messages via SIM-based gateway integrations.

• Small Office PBX Extension — Use TG200 as a SIP registrar or trunking gateway if you don’t have a full IP PBX.

Why Choose Yeastar TG200 VoIP?

• Delivered with a 3‑year warranty, backed by expert regional support.

• High compatibility with existing SIP systems and plug‑and‑play deployment.

• Powerful call features like callback, blocklist, auto CLIP, route balancing, time-based routing, and detailed call logs. User-friendly GUI enables fast configuration.