FreeSwitch Jobs
FreeSwitch is a free, open-source application server for real-time communication. It is used to build PBX (private branch exchange) systems, video conferencing and IVR (interactive voice response) services. The server is supported cross-platform, and is used with MacOS, Windows, Linux, Solaris and multiple BSD (Berkeley Software Distribution) descended operating systems of which FreeBSD is the most popular.
FreeSwitch is licensed under a free software license, giving a recipient the ability to adapt and redistribute the software. There are still restrictions, as FreeSwitch is not in the public domain.
Hire FreeSwitch Developerstrying to get a simple wss://[login to view URL] up and working and proxying to SIP server. i have [login to view URL] built out but i might be missing something so small need someone to help troubleshoot to get it working.. i have [login to view URL] 1.) use [login to view URL] or [login to view URL] hopefully to verify webrtc client demo (login with sip:1111 @ [login to view URL]...
I need this built in laravel of the newest kind. the site features should look like: [login to view URL]
LUA script I'm looking for a IVR LUA script : Step 1: Play Sound (What is your employee number Step 2 : See in Mysql all employee number (active) Step 3 : Load the employee in the IVR Step 4 : The call enter the employee number Step 5 : The script verify in Mysql if the employee are In or Out Step 6 : The system put the employee out if are in and put in if are out in the MySQL Kind Regard
mini POC WEBRTC-to-SIP architecture backend buildout for SIP server to webrtc (kamailio or janus) docker build setup to Sample this SIP-WEBRTC (to enable websockets) client= webrtc sip register=SIP server does RTP have to flow through kamailio or SIP-WEBRTC as well trying to model off this [login to view URL]
we want to separate Freeswitch server on our bigblue button sever
Enable TLS port 5061 on fusionpbx System installed on google cloud
