WebRTC phone recognizes the sip extensions but doesn't go online
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The webRTC phone is recognizing the sip extensions I have created, it recognizes that it is one of the extensions in this case the extension "2001" but it's not appearing online. I followed the instructions from this link https://www.fop2.com/docs/webrtc_guide.php but in my case with a certificate created with letsencrypt and it continues to be offline. I tried it using the default asterisk certificate and it continues to stay offline. I'd really appreciate some feedback on this and if someone has had this issue before, how did you manage to solve it?
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