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Asterisk full GUI PBX with SBO/RBC VoIP Call Bandwidth Optimizer for CALL BLOCK AREA VOIP OPERATION AND also need call filteration system

PLEASE DON'T BID IF YOU DON'T KNOW WHAT IT IS ABOUT send query before bidding

Server X = Asterisk Server

Server Y = Window based Asterisk Client server

Main goal minimize Bandwidth in client side with quality voice .

Required bandwidth compression (upto 60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port

Explanation of scenario:

1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y.

2. Server Y {Windows based Asterisk server with PRIVATE NETWORK IP, receiving calls from server X and sending to gateways (quintum, goip etc brand) or E1 cards. (Please take a look at this page for better clarity [url removed, login to view])

3. Number of Server Y can be unlimited.

4. Number of Gateways/E1 cards per server Y can be unlimited

5. For server Y installation will be window based software based on Asterisk can be easily downloaded from our website and can be installed must be in [url removed, login to view] file mode ([url removed, login to view] for reference). This installation file will require specific activation code from us to activate otherwise will not run. And each time this file run must register with our Server X to see if the activation code has been expired or not. We want our client side to be based on windows not something on USB flash dirve linux.

6. Server X must have codec conversion i.e., can change the voice codec of SIP G729 to SIP G723r63 or visa verse

7. Server X to Server Y voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination this is very important as this service will mainly be used for voip block countries. we will use:

- iax trunks in trunking mode.

- Open vpn static mode and dynamic mode

- Tnic static and dynamic mode

8. We need

- Commercial class Asterisk web billing

- GUI for adding gateways

- Adding client

- Setting Dialing plan /Prefix

- viewing active calls,

- Real time billing cdr so that can be use to bill the call originator

- Localization of the cdr as per time frame of the originator to match the billing of the originator

- Real Time ASR, ACD per Gateway, Per IP, Per destination based on 15 minutes/ 30 minutes/ 45 minutes/ 60 minutes, daily, weekly and monthly basis can be used set standard to block call below a certain level of ASR and ACD. To satisfy the carrier to fulfill a certain level of quality assurance.

- Must able to generate error 503 for SIP Protocol when the termination side is down to assimilate the call generation side to stop sending the call

- Must have multi-user login so that user of different client of server Y can log in and set their own parameters such as client, codec, destination, origination, protocol, gateways, dialplan, price, costing, billing etc

- we will provide Dedicated server asterisk and client asterisk

- Require to configure IAX trunking, BW compression can be measured making the SIP-> IAX call trunking, need a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW)

- Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes;

- require continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may servers.

- continue working on project by building up WEB interface for main server adding Billing, adding IAX trunks and other options for user friendly GUI

- WE REQUIRED THE SAME SERVICE AS [url removed, login to view] AND/OR [url removed, login to view]

ALSO call filtration system which can limit number of caller per port.

Beceriler: Asterisk PBX, Cisco, Linux, Ağ Yönetimi, VoIP

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İşveren Hakkında:
( 0 değerlendirme ) United Arab Emirates

Proje NO: #15552342

6 freelancers are bidding on average $676 for this job

seekdeveloper

Hi, I have read your post and understood your requirement. Looking for the freelancer to work on your next project? Or just need some issues/bugs/fixes ASAP? Relevant Skills and Experience • Asterisk PBX, VoIP, Cisco Daha fazlası

in 10 gün içinde777$ USD
(2 Değerlendirme)
5.0
openasterisk

Hi, Trust you are doing well I am confident enough to fulfill your requirements as I have done similar types of work before. Relevant Skills and Experience I am an expert with Asterisk PBX, Cisco, Linux, Network Adm Daha fazlası

in 10 gün içinde555$ USD
(6 Değerlendirme)
4.2
shobhanasaxena

Hi, On behalf of CS Infotech having experience of 7+ [url removed, login to view] have a team of experienced developers & designers who are capable of completing this project on time with quality. Relevant Skills and Experience We have g Daha fazlası

in 30 gün içinde722$ USD
(0 Değerlendirme)
0.0
AzzkaNoor

I have read your project details. I have to ask a few questions. Can you please message me via chat so we can discuss all the details to elicit all the requirements and hence start the development? Relevant Skills and Daha fazlası

in 10 gün içinde555$ USD
(1 Değerlendirme)
0.0
eugenialettonia

I have already done a similar project Server A receives calls from a Trunk SIP forwarding to a SERVER B in IAX The Server B have a dongle GSM Competenze ed esperienze rilevanti Asterisk and PHP Pietre Miliari propost Daha fazlası

in 15 gün içinde722$ USD
(0 Değerlendirme)
0.0
nullanelweb

The project is possibile configuration. I have made a configuration similar. I want know the HW Competenze ed esperienze rilevanti GSM VOIP ASTERIX ROUTING for COUNTRY Pietre Miliari proposte $722 USD - GSM VOIP ASTE Daha fazlası

in 10 gün içinde722$ USD
(0 Değerlendirme)
0.0