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hfvoip2012

asterisk,linux ,php

Username: hfvoip2012

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Location: Shanghai, China

Member since: July 2012

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5.0/5

(1 reviews)

2.6
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  • $192 USD
    5.0
    Profile image for Seller mattl

    mattl

    May 22, 2013

    Yang is the best Asterisk/ VOIP developer I have worked with. He gets everything that I say, and works very quickly. I truly value him as a member of my team and will be working with him very frequently on a ongoing basis. He has already created several AGI scripts, customized Vici dial, and created some really unique dial plan solutions for us.

    Project Description:We are using the vici auto dialer platform and need a custom agi script created for it. Vicidial allows you to press 1 and then connect to a live agent, or an extension. I figure the best way to connect...
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  • ₹600 INR Sep 25, 2013

    supply hardware server hardware , install and maintain (1 year support with response time commitment) vicidial in shanghai area for 60 agents. two or 3 custom IVR scripts to be deployed.Pls revert if you are interested or any one in u r circle can take up this task

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  • $1666 USD Jun 5, 2013

    Total Solution NeedFor a IVR Telephone Dating Service Program1 Multiple callers are connected simultaneously2 Callers choose who they want to talk with3 Voice profile setup capability4 Can detect area codes and switch lines5 System must be easy to manage and adjustexample sites: http://www.matchandtalk.com/Do you understand this business?

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  • $111 USD May 28, 2013

    I am searching for a Vicidial expert to help me setup my Vitelity trunk. I have the trunk setup but carrier NO ANSWER status are being ignored by vicidial. So need a asterisk expert to troubleshoot problem and see what is going on.Example of problem Trunk disposition = NO ANSWER, Vicidial Disposition stays on NEW user VDAD. On the outbound real time trunk report "AST_timeonVDAD.php" these numbers also timeout on 2 minutes even though my dial time is 27 seconds, so it seams that the Trunk returns a NO ANSWER and vicidial does not know what to do with the call and it waits the full 2 minutes to time out, and lead stays on NEW but its been dialed several times.

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  • $330 USD Apr 13, 2013

    We have server (up & running):- Dynamic IP (with ddns host)- SSH available from outside- Ubuntu 12.10- Running (no configuration) asterisk (apt-get install asterisk ...)- TDM400P card (4x FXO)- 6 core AMD pc# Desired infrastructure #3 x incoming land lines + 1 Skype Connect (SIP) account with few (~4) numbers.5 x local SIP accounts for attending (In future more, maybe 10)Features we need:- Calls transferring- MOH (music on hold)- On incoming call ring on 3x sip accounts, if all busy play message (all lines are busy, please wait or call later...)- After hours message- Digital fax support- For outgoing calls use random line 1 or 2, separate extension for skype outgoing- Important: on incoming / outgoing call make curl request (for logging, we like to have in mysql DB all incoming and outgoing call)In separate order we might request for:- Call recording- Adding FXS/FXO lines (currently we have no hardware and no PCI port available)

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