Example, Incoming external call (party A) is answered by party B (normal sip phone). If party B makes an attended transfer to an WebRTC extension (party C) you see the CallerID from party B in WebRTC. After Party B completed the transfer the CallerID in WebRTC is not updated to party A. Normal Yealink sip phones update the CallerID offered by the PAI Header, it would be nice if the WebRTC clients do the same.
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It seems that Asterisk doesn’t notify when a session is replaced (attended transfer), please see the following post: Asterix and sip "replaces" header support - #9 by balzor - Asterisk Support - Asterisk Community
In our tests, when completing a transfer, asterisk does not always send a reinvite with the updated Remote Identity/P-Asserted-Identity,
Could you please confirm?