Update CallerID after attended transfer

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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