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VoIP GSM Gateway for Cost Savings

The NeoGate TG series is a series of VoIP GSM gateways that connect GSM networks to VoIP networks, allowing two-way communication between GSM and VoIP (and vice versa). The gateways range from supporting 1 to 16 GSM channels and provide cost savings over mobile-to-mobile calls while serving as a backup when landline networks fail. All models support common VoIP protocols and functions and are easily configured through a web-based interface.

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66 views2 pages

VoIP GSM Gateway for Cost Savings

The NeoGate TG series is a series of VoIP GSM gateways that connect GSM networks to VoIP networks, allowing two-way communication between GSM and VoIP (and vice versa). The gateways range from supporting 1 to 16 GSM channels and provide cost savings over mobile-to-mobile calls while serving as a backup when landline networks fail. All models support common VoIP protocols and functions and are easily configured through a web-based interface.

Uploaded by

l3st4t7
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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TG Series

NeoGate TG series is a series of VoIP GSM gateway connecting GSM Network to


VoIP Network directly, which can support two -way communication: GSM to VoIP
or VoIP to GSM.
It is the best solution ever to connect IP - based telephone systems, and
softswitches to GSM network ; and also the best fallback solution w hen landline
goes down.

Cost saving

Save

phone

bills

tremendously

with

mobile-to-mobile calls

Back up

Easy to install

Everything can be easily set up in the Web


based management interface

Easy to integrate

Work as a cost-effective backup when the

High compatibility with major IP-PBX and

landline network goes down

softswitch brands

TG100

TG200

TG800

TG400

TG1600

Mobile: 1/2/4/8/16 GSM Channels

GSM Frequency: 850/900/1800/1900MHz

LAN: 1 (10/100Mbps)

SMS Sending and Receiving

Antenna splitter (4 in 1): TG400/800/1600


Protocol: SIP, IAX2

Send Bulk SMS

SMS Center

SIP Trunk: Support

Gain Adjustment

SIP Peer Mode

USSD

SIP server for IP phones included

PIN Modify

SIP Response Code Switch

Carrier Selection: Auto/Manual

Session Timer

Balance Alarm

Call Progress Tone Generation

Caller ID/CLIR

1 Stage/2 Stage Dial

Open API for SMS and USSD

Transport: UDP, TCP, TLS, SRTP

Voice Codec: G.711 (alaw/ulaw), G.722,


G.726, G.729A, GSM, ADPCM, Speex

Network Protocol: UDP, TCP, TFTP, HTTP, SSH

Echo Cancellation: ITU-T G.168 LEC

NAT Traversal: Static NAT, STUN

DTMF Mode: RFC2833, SIP INFO, In-band

Static Route

OpenVPN

Firewall

Calling Type: Termination (VoIP to GSM),

VLAN

Origination (GSM to VoIP)

QoS/ToS

GSM Ports Group Manage

DDNS

VoIP Trunk Group

Configure backup/restore

Incoming /Outgoing Routing rules

Firmware upgrade by HTTP/TFTP

White List and Black List

Web based configuration

Hotline

Call Duration Limitation

Call Transfer

Dimensions

Call Waiting

(mm)

Call Back

Call Detail Record (CDR)

Power Supply
AC 100~240V/50~60 Hz

TG100

110x70x24

Call Status Display

TG200

213x160x44

100-240V 50/60 Hz 0.5A MAX

NTP

TG400

213x160x44

100-240V 50/60 Hz 0.5A MAX

IP Blacklist

TG800

340x210x44

100-240V 50/60 Hz 1.5A MAX

Network Attack Alert

TG1600

440x280x44

AC 100-240V 50/60Hz

Packet Capture

System Logs

Real Open API Protocol (Based on Asterisk)

(DC 12V, 1A)

Operation Range: 0C to 40C, 32F to 104F

Storage Range: -20C to 65C, -4F to 149F

Humidity: 10-90% non-condensing

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