VOIP Software Jobs

8 were found based on your criteria

  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    I needs someone to setup my SIP trunk with my FreePBX and get it to work. I've bought my SIP Trunk at DIDlogic.com. I want to call from the browser and wants to know what to do to make it work. Is it enough with a SIP Trunk and the iSymphony module? Or do I need hardware? If I need hardware, can it be outsources? Best regards
  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    Hello, Please include the word "A2Billing" at the top of your application so I know your human please. We are looking for a very simple VOIP Service Platform with Web Interface. (function over style) NOTE: Please only bid if you have proven experience with A2Billing & didx / didlogic and if you have done something similar. Possible intergation with didx API or open to other suggestions. Want to build simple DID website. The system will be using A2billing and Asterisk, which we ...
  • Hourly – Less than 1 month – 30+ hrs/week – Posted
    Develop a software driver to interact with a Highway Advisory Radio via a wireless cellular VOIP connection. Requires some knowledge of cellular, voip and cloud-based voip using a RESTful API. Driver to be written in .net (likely C#) on a windows platform. Need a 2-3 week turnaround. Some R&D likely
  • Hourly – More than 6 months – Less than 10 hrs/week – Posted
    This is a Long term opportunity for a real expert who know Free Switch and ASTPP. We have already a calling card platform and now we looking for an expert who can do below : 1) Installation Free Switch in new server with all market standard codec. 2) IN and Out Trunk setup to call route out all over the world like voxbeam gateway and capability to terminate call 3) Install and configuration ASTPP on top of Free Switch. 4) Design ...
  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    see the attached documents, i am looking at scenario #6 for now with some REST API setup and some demo code for webrtc running looking for the server to listen to port 443 instead of the usual 3479 coturn server= https://code.google.com/p/coturn/ the server wil be behind firewall/NATed
  • Hourly – 3 to 6 months – 10-30 hrs/week – Posted
    Modulis is a leading VoIP software development company in Canada. We developed a telecommunication platform based on OpenSIPs / Asterisk with a web interface in Ruby/EXTJS. We are looking to hire an experienced project manager to manage a team of up to 5 developers. Your duties: -Interview developers and tests them -Hire and negotiate terms with developers -Train developers and get them started with the development environment -Ensure our developments standards are met -Review code -Keep an in-depth understanding of ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    We have a video uploading that explains the obstacles and future opportunities. http://youtu.be/SN-VdYOhmKY In the meantime, I'll describe the two problems we are having: 1. Dialing Zero "0" Says it's an invalid extension. 2. The dial by name directory doesn't work as well. Hopefully someone can shed some simple light on these challenges. Respectfully, Kyle O'Brien
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