Supplier of 8 Channels GSM VoIP Gateway For Call Termination

Joined: Fri 25 of Mar, 2011

Supplier of 8 Channels GSM VoIP Gateway For Call Termination

Posted:Fri 25 of Mar, 2011 (08:25 UTC)
Link To Buy:


Detailed Product Description

8-Channel GSM VoIP gateway
Support SIP/H.323;
GSM 850/900/1800/1900Mhz;
NAT Transversal;
VLAN and QoS...

Basic Functions

1.To realize the uplink and downlink calls between the GSM network and the VoIP network, eliminating the distance roaming charges.

2.To transmit the caller number from the PSTN to the VoIP.

3.To minimize the risk of charge losses via bi-directional password authentication (call authorization) and trust list authentication.

4.To meet special requirements of various call forwarding.

Key Features

Provide 8 cellular channels for IP-PBX

Support open standard SIP Protocols (IETF SIP V2)

Support SIP proxy mode

Multiple GSM VoIP-8 grouping mode

Two 10/100 Ethernet ports for the LAN and an additional device

Guad band GSM module: support GSM 850MHz, 900 MHz, 1800 MHz, 1900MHz

Speech quality ensured by QoS at LAN, IP layers and comprehensive jitter buffer

VLAN and QoS support

NAT Transversal

Voice prompts, HTTP Web

Highly stable embedded Linux operating system in high performance ARM 9 Processor

Enhanced Features

LEDs for Power, Ready, Status, WAN, PC, GSM

Call forward from GSM to VoIP and VoIP to GSM

Dial in mode or dial out mode only

Dial Plan

Password protection for both GSM dial in or dial out

Retransmit GSM Caller ID to VoIP terminal

Dynamic selection of codecs

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)

Firmware upgrade from GUI

Supported Standards

ITU: H.323 V4, H.225, H.235, H.245, H.450


RFC 2327 - SDP

RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

RFC 2976 - SIP INFO Method

RFC 3261 - SIP

RFC 3264 - Offer/Answer model with SDP

RFC 3515 - SIP REFER Method

RFC 3842 - A Message Summary and Message Waiting Indicator

RFC 3489 (STUN) - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)

RFC 3892 - SIP Referred-By Mechanism

Proprietary Firewall-Pass-Through Technology

Codec: G.711 (A/µ law), G.729A/B, G.723.1

DTMF: RFC 2833, In-band DTMF, SIP INFO

Web-base Management

PPP over Ethernet (PPPoE)

PPP Authentication Protocol (PAP)

Internet Control Message Protocol (ICMP)

TFTP Client

Hyper Text Transfer Protocol (HTTP)

Dynamic Host Configuration Protocol (DHCP)

Domain Name System (DNS)

User account authentication using MD5

Hardware Specifications

Processor: ARM9E 133MHz

DSP: VPDSP101-4 100MHz

Memory: RAM 16MB/ Flash 4MB

GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz

Power: Input AC100V ~ 240V, output 12 Vdc 2000 mA

Power consumption: 5W maximum

Operating temperature: 10°C to 40°C (32°F to 104°F)

Storage temperature: 0°C to 50°C (32°F to 122°F)

Working Humidity: 40% ~ 90% Not congealed

Weight: 1550 g (3 lb) (Including AC/DC Adapter)

Warranty: 1 year

In particular, the GSM VoIP gateway supports multi device groups. With its low price, excellent voice quality, and powerful features, the GoIP series gateway is the first choice for system integrators, traffic operators, and softswitch manufacturers.

Packaging Details
Unit Type: piece
Package Weight: 1.5kg (3.31lb.)
Package Size: 30cm x 15cm x 5cm (11.81in x 5.91in x 1.97in)

Add: 11B, Palm Building, 863# Jie Fang Bei Road, Yue Xiu District, Guangzhou, China
TEL: +86 20-83233345 FAX: +86 20-83233347
Skype: lvthea
Website: ;