Is it possible to create a PJSIP extension that can be used by both a webrtc client (not the built in webphone) AND a regular sip-phone?
I can’t speak to FreePBX but from an Asterisk perspective it isn’t possible to have a single endpoint. This is because while WebRTC uses SDP it requires extensions and formatting that renders it unusable by normal devices. While on incoming calls to Asterisk you can deduce which one to use - outgoing becomes more difficult as you can’t offer both. Noone has as of yet spent any time on solving that problem or trying to make it more intelligent.
To that end within Asterisk it requires configuring 2 endpoints in the end with different names.
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