Yeastar TG200 2*GM1

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Yeastar TG200 2*GM1

SKU: yes-tg200

Availability: Special Order - Call for Pricing

The Yeastar Neogate TG200 is a GSM gateway for connecting GSM networks to VoIP networks for two-way communication. The TG200 is ideal for connecting IP Phones, soft switches, and IP-PBX's to GSM network, and is the best fallback solution for landline outages.

DESCRIPTION

Manufacturer: Yeastar


Yeastar TG200 VoIP GSM Gateway connects GSM or WCDMA to VoIP networks to provide two-way communication: GSM/WCDMA to VoIP and VoIP to GSM/WCDMA. This allows you to connect most IP-based telephone systems including Yeastar IP Phone Systems, and softswitches to a GSM or WCDMA; which can provide a sophisticated fallback solution when landlines go down, or be used to increase call traffic capacity by providing additional dial-tone.



Yeastar TG200 Compatible With:


  • Asterisk
  • Microsoft Lync
  • Trixbox
  • Elastix
  • FreeSWITCH
  • 3CX
TECHNICAL INFO
SPECIFICATIONS
Number of Ports 2
GSM Frequency 850/900/1800/1900 MHz
WCDMA Frequency 850/1900 MHz, 850/2100 MHz, 900/2100 MHz
4G Data -
4G LTE Band Depends on the module type
Protocol SIP, IAX2
Antenna Splitter (4 in 1) Supported
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 -
Network Protocol FTP, TFTP, HTTP, SSH
LAN 1 10/100 Mbps Ethernet Interface
NAT Traversal Static NAT, STUN
Network DHCP, DDNS, Firewall, OpenVPN, Static IP, QoS, Static Route, VLAN
Power AC 100-240V
Operating Range 0°C to 40°C, 32°F to 104°F
Storage Range -20°C to 65°C, -4°F to 149°F
Humidity 10-90% non-condensing
Mounting Desktop, Wall-mount
Compatibility Interoperable with Asterisk, Lync Server, FreePBX and certified with Elastix
Features
  • 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
  • Whitelist and Blacklist
  • 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