Hotgsm Voip Goip Gateway Sip Trunk To Asterisk Ip Pbx For Sale

GoIP VOIP Gateway GSM Converter

SIP IP Phone Adapter

Quad Band 1xGSM SIM Gateway:

Trunk to Asterisk iP PBX





Note:We provide different types for GoIP ,VoIP gateways 1,4,8 channel,IP phones,FXS1,2,4,8 ports ,FXO2,4 etc , if you are interested in , please visit our store--taobaotown,or contact us via message,thanks!


Free roaming:


In order to promote our product ,Now we have added some new funcitions in GoIP.By adding the new funcitions,you can build the call without using the softwhich or platform,just depending on the internet.it makes our calls more convenient and easier.what’s more, it can save a lot of call fee.Here is the following example:

Example:


peer to peer:it is a new function that you can use our voip without platform,what is more,it is free roaming for international call.you just pay the local call

Model 1

Model 2

Model 3

Model 4





 Applications:

 

 

The basic use of our GOIP/VOIP as follows:

PSTN->VOIP,

 

VOIP->PSTN,

 

 

Telephone is a common phone. Softswitchis software installed in a computer. It looks upon as a server, Goip looks as client. The scheme needs goip、telephone、
softswitch、another phone login softswitch through common Gateway device. Then you can PSTN->VOIP and VOIP->PSTN

 

Call Back

 

 

Summary:

 

GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cell phone roaming via the VoIP network.



GoIP GSM Gateway is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks.

 

GoIP GSM Gateway is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GoIP GSM gateway also provides significant savings in usage (long distance or international), infrastructure and maintenance cost compared to conventional PSTN.

 

Benefits of GoIP:

Extensive product compatibility with industry leading vendor

Cost-Savings on phone calls between mobiles or to PSTN

Easy to install – IP device with Web based management interface

Can be managed and monitored remotely over Internet - a great service to offer to customers by system integrators

GoIPs can be grouped together to establish GSM gateway cluster

Termination between GSM/VoIP

Schedule or on-demand SMS Broadcast messages to users

(Additional SMS server is required)

 

Key Features:

Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)

Single Server Registrations

SIP peer-to-peer mode

SIP proxy mode

GSM module – 850MHz, 900 MHz, 1800 MHz, 1900MHz

Advanced jitter buffer

VLAN and QoS support

NAT Transversal

Call forward from GSM to VoIP and VoIP to GSM

Password or Trust list protection for dial in mode and dial out

Comprehensive dial Plan

Retransmit GSM Caller ID to VoIP terminal

Dynamic selection of codecs

Echo cancellation for Speakerphone

Comfort noise generation (CNG)

Voice activity detection (VAD)

Firmware upgrade from GUI


Basic 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

Enhanced Features:

Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
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 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

 

 

 

 

Payment


»Only Paypal are accepted!
»check payment will be held until the payment is completed cleared on seller side.
»We prefer payment donewithin 3 daysafter the Winning offer
 

»this item can be used in the worldwide!
»Delivery address on Paypal Only--We only ship to the given address in Paypal payment.Please make sure payment address inPaypal is correct. Any non-received items caused by invalid address registered on Paypal are not in our full refund or replacement policy.

»If you want to speed up shipmentby (EMS,DHL,FEDEX,UPS and etc), please contact me for specific shipping rates.

 

Returned Policy


»We will refund as soon as we receive your returned item (items must be in their original condition). Please kindly note: The shipping cost will NOT be refunded, and you need to pay the shipping cost to send the items back (except the item is defect).
 

Service


»if you want to buy many device,we will provide you a good discount to you and if you have any other questions, pls feel free to contact us at any time! We will reply within 24 hours.

About response


»Positive response will be left after receiving positive response from buyers
»Before leaving any negative or neutral response, please give mea chance to assist you in any problems that you may encounter. I will make every effort to satisfy you.
 

Other Terms

»Please use this item properly and respect other people's privacy.
»Size and weight are approximate only.


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Hotgsm Voip Goip Gateway Sip Trunk To Asterisk Ip Pbx

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Hotgsm Voip Goip Gateway Sip Trunk To Asterisk Ip Pbx:
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