Product Name: Trunking GatewayModel Number: GYXS100-TGPlace of Origin: ChinaGYXS100-TG trunking gateway is one of the products in VoIP product series.
GYXS100-TG performs IP to PSTN interconnection function by packetizing and
depacketizing voice calls between IP and PSTN. It is widely used for PSTN
termination for VoIP calls, or providing PBX access to IP networks.

GYXS100-TG is specifically designed for service providers and large enterprises
to provide VoIP services. With 1U form factor, a GYXS100-TG is equipped with a
main control module, two line module slots (for one or two 4-port T1/E1 modules),
and two power supply slots (for one or two AC or DC power supplies).

Voice processing capability is the most important requirement and measurement for
all trunking gateways. GYXS100-TG configured with one to four T1/E1 interfaces
can maximally process 120 simultaneous VoIP calls. GYXS100-TG embeds powerful
DSP technology which possesses 4800 MIPS processing power, for full load voice
processing to support CODEC (G.711, G.729A, G.723), echo cancellation (G.168),
DTMF relay (RFC2833), and fax relay (T.30, T.38).

GYXS100-TG supports many types of features to satisfy various application
requirements.

Features include calling/called party number translation, routing selection,
RADIUS billing interface, second stage dialing and ringback tone, DTMF auto
dial, tone detection, RTP voice proxy (for NAT traversal), T.38 fax relay.
GYXS100-TG supports ISDN PRI and SIP protocols and it is the ideal PSTN
termination trunking gateway for IP phones, PC phones and PBX bridge to
VoIP service.

Like other VoIP products, GYXS100-TG has the Web GUI and text configuration
interfaces.

In addition, a user can remotely upgrade, maintain, download usage logs from
and collect statistics from an installed GYXS100-TG by using EMS system.

Compared to other similar products on the market, GYXS100-TG embeds newest
VoIP technology and standards, and is an open platform to support many new
telecommunication applications to come.

Specifications:
1) Flexible configuration of one to four T1/E1
2) Toll voice quality
3) Guarantee 120 DS0 voice processing with any CODEC
4) Ideal PSTN termination for IP Phones, IP PBX and PBX access
5) Embedded with RTP voice proxy, digit translation, auto dial and RADIUS
billing interface
6) New technology including SIP protocol, frequency domain echo cancellation
and iLBC CODEC