[asterisk-dev] Asterisk does not want to switch back to voice mode after successful T38 passthru faxing from SIP-exten to SIP-trunk

[Evgen]

Asterisk does not want to switch back to voice mode after successful T38 passthru faxing from SIP-extension to SIP-trunk.
LOG: "chan_sip.c:14232 handle_request_invite: RTP re-invite after T38 session not handled yet !"
Is there any information when it will be handled?