8 GSM SIM Channel GoIP Gateway TGG800

La description

VoIP GSM Gateway TGG**0 are VoIP GSM Gateway for call termination (VoIP to GSM) and origination (GSM to VoIP). It is SIP&H.**3 based and compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster. It can enable to make 8 calls simultaneously from IP phones to GSM networks and GSM networks to IP phone.

Key Features

8 GSM channels, up to 8 SIM cards
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP & H.**3 protocol, Communicates with other gateway or PC
Quad band, IMEI changeable, Remote Access
Support of SMB*6 SIM Bank
Optional SMS termination
Allows your program send/receive SMS with AT command
Easy to install and administrate
Auto Balance and Recharge
Auto BTS changeable

Support one stage dialing
Support free mode-two stage dialing and assigned mode-one stage dialing
Call Back feature
All functions can be set on web
Provide CDR
LEDs for Power, Ready, Status, WAN, PC, GSM
Dial in mode or dial out mode only
Call forward from GSM to VoIP and VoIP to GSM
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codec


Advanced Features


Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Echo cancellation for Speakerphone
Comfort noise generation (CNG)

Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese


Network Features


ITU: H.**3 V4, H.**5, H.**5, H.**5, H.**0
RFC ***9 - RTP/RTCP
RFC ***7 –SDP
RFC ***3 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC ***6 - SIP INFO Method
RFC ***1 – SIP
RFC ***4 - Offer/Answer model with SDP
RFC ***5 - SIP REFER Method
RFC ***2 - A Message Summary and Message Waiting Indicator
RFC ***9 (STUN)- Simple Traversal of UDP Through Network Address Translators (NATs)
RFC ***1 - SIP “Replaces” Header
RFC ***2 - SIP Referred-By Mechanism
draft-ietf-sipping-cc-transfer**4 - Session Initiation Protocol Call Control Transfer
Codec: G.**1 (A/µ law), G.**9A/B, G.**3.1
DTMF: RFC ***3, In-band DTMF, SIP INFO
Web-base Management
PPP over Ethernet (PPPoE)
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP Client................................