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Is it possible to make an analog of webrtc2sip from FS/Asterisk or something stable?
Raised webrtc2sip, everything seems to be fine, but I'm not sure about its stability. Is it possible to make an analogue from FS / Asterisk, etc.?
The client is sipml5. I tried oversip, but for some reason there was no voice. Very interested in real experience.
Sipml5 is already built into the site, so I want a solution for which I would not have to change the js code.
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