Freeswitch Jobs

33 were found based on your criteria

  • Fixed-Price – Est. Budget: $200.00 Posted
    We need to modify / improve the "Call History" module in the Kazoo-UI for our 2600hz Kazoo stand-alone private cluster. The main issue our customers have is that they want to see only the unique calls for each of their agents. The current Call History module shows all the details of each leg of each call... Almost all calls have 2 results, some have 10+ results. There should be a way to group the call leg details under related UUIDs, which ...
  • Hourly – 1 to 3 months – 30+ hrs/week – Posted
    We have multiple telco carriers on bound to carry the calls and I assume this platform will be built on Free Switch. A responsive UI template will be provided to you to have this built into. Please let me know if you have any questions and I would also need an estimate on pricing as well as a time frame until completion. 1) We upload a list to the system with contact phone numbers. 2) We also upload two different ...
  • Hourly – More than 6 months – 30+ hrs/week – Posted
    We are seeking someone with good knowledge of freeswitch that also knows NewFies dialer. The requirements are as follows Ability to understand server and troubleshoot on NewFies Platform. We are only seeking someone seeking full time employment that wants to grow with us. Although we make several million dials daily, the FreeSwitch/NewFies dialer is not in production, however we want it to be reliable and stable for everyday production.
  • Hourly – More than 6 months – Less than 10 hrs/week – Posted
    We need ongoing assistance with a freeswitch deployment. We are moving from Asterisk to Freeswitch on some platforms and need someone ongoing to look and help with our dialplans.
  • Fixed-Price – Est. Budget: $250.00 Posted
    I have a Barix Instreamer that is able to push audio to icecast. When the stream is live I need icecast to connect to a SIP endpoint and have the audio transmitted via G722 to that sip endpoint. When the icecast stream is disconnected the sip connection needs to be taken down. I believe this can be done using mod_shout in FreeSWITCH and the on-connect and on-disconnect functionality of icecast. I have FreeSWITCH and Icecast installed and running.
  • Hourly – More than 6 months – Less than 10 hrs/week – Posted
    We're getting some freeswitch errors when phones try to register. Using a regular SIP client like X-Lite works fine and registers, but using a Polycom VVX 500 fails. We have a Kazoo deployment of FreeSwitch with kamailio in the front. Give us a hand :)
  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    We need some basic configuration of freeswitch which we would like to extend with additional features - Disable proxying global - Tune T.38 support - Capacity management based on data from external system (via SIP 3xx messages) - CDR management enhancements
  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    Voice & Web App Developer We are looking for a developer to continue building an application built with JavaScript and Adhearsion. This developer will work closely with an experienced architect to produce high-quality code with cutting edge technologies and proven best practices. Must be experienced with both telephony and web programming. Required skills: Twitter oembed Nexmo API JavaScript frameworks (such as Meteor.js, Angular.js, Backbone.js etc. Not just jQuery) HTML5 and CSS Adhearsion Version control Automated deployment and/or ...