asterisk voip gsm gateway 4 port

 

Key Features

Open Standard VoIP Protocols (SIP&H.323)

Single or Multiple Server Registrations

Two 10/100 Ethernet for WAN / LAN connections

Peer-to-Peer IP Calls

Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer

Line Echo Cancellation

VLAN and QoS support

NAT Transversal and Router functions

Voice prompts, HTTP Web, Auto Provision support for configuration and updates

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

 

Basic Features

LEDs for Power, Ready, Status, WAN, PC, FXS
Dial in mode or dial out mode only

Call forward from GSM to VoIP and VoIP to GSM

Dial Plan   gateway gsm

Retransmit GSM Caller ID to VoIP terminal

 

Enhanced Features

Dynamic selection of codec

Advanced jitter buffer  

Automatic traversal of NAT and firewall

VLAN / Qos

Router   gateway gsm

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

 

Supported Standards

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

RFC 1889 - RTP/RTCP

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

RFC 3891 - SIP "Replaces" Header

RFC 3892 - SIP Referred-By Mechanism

draft-ietf-sipping-cc-transfer-04 - Session Initiation Protocol Call Control  Transfer 
Codec: G.711 (A/µ law), G.729A/B, G.723.1

DTMF: RFC 2833, In-band DTMF, SIP INFO 
Operating temperature: 10°C to 40°C (50°F to 104°F)

 

Physical and Environmental

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

Power: 12 VDC 2A (110V-220V) (AC/DC adapter included)

Warranty: one year

 

FAQ:

1. What is the communication network can be supported by GoIP?

GSM network

2 What is the software can be supported by GoIP?

VOS,VPS,VoIP switch,3CX,P·B·Xes,free PBX,MOR

3.What is the bandwidth requested by GoIP?

It is up to your voice code.G723 need 23kbps per channel,G729 need 34kbps,G711 need 90 kbps

4. What is the hardware can be supported by GoIP?

 IPPBX and so on

5.How to send the devices to customers?

We can send products by using the international logistics, including

  DHL, UPS, TNT, EMS,  Fedex, ARAMEX.

We can promis you that we will delivery the good to its destination effectively and safely.

 

Our company:

 YX International Information Co., Ltd is a manufacturer of GOIP Gateway, CDMA gateway, VOIP Products,  GSM Modem ,Voice Recorder and IP Phone with well-equipped testing equipment and strong technical force. Which is found in 2007, with almost 8 years export experience, has won a good corporate reputation in foreign and domestic market

  Through the effort of our entire staff, currently, YX Co., Ltd has already exported to Bangladesh,America, European, Middle east, South Aisa, Southeast Aisa and 20 other countries and regions.

 

Our service:

we are manufacturer and can give you factory price.

We provide you stronge and free technical support about VOIP server and A-Z termination. (6 days*12 hours)

We can help you to rent a VoIP server in USA.

We can provide you copy version VOS.

We can install Relay server, SMS server,SMB server for you.(for free)

We can offer voip terminal guide.

We may need your termina routes.

We offer full technical support untill your devices work well.

     

Any inquiry, feel free to contact me. 

Skype: ericazhong2

Tel: +86-7522777557

Fax: +86-7522159233

Web: www.yx19999.net

Mobile Phone: 8615112198908