[Serusers] Meaning of and proper response for a 401

There's a UA in our network that responds to a reinvite with a 401. U PROXY.IP.ADDRESS:5060 -> UA.IP.ADDRESS:63718 INVITE sip:1101XXXXXXX@UA.IP.ADDRESS:63718 SIP/2.0. Record-Route: <sip:PROXY.IP.ADDRESS;ftag=as7bbe322e;lr=on>. Via: SIP/2.0/UDP PROXY.IP.ADDRESS;branch=z9hG4bK9159.cd2a95b4.0. Via: SIP/2.0/UDP PSTN.GW.IP:5090;branch=z9hG4bK39c17f01;rport=5090. From: <sip:13154476314@our.domain.com>;tag=as7bbe322e. To: "User One" <sip:1101XXXXXXX@our.domain.com>;tag=202d9942. Contact: <sip:13154476314@PSTN.GW.IP:5090>. Call-ID: 6b3f44351f5fab945ac212a2141df93a(a)UA.IP.ADDRESS. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 314. P-hint: LR. . v=0. o=root 15241 15243 IN IP4 216.143.130.90. s=session. c=IN IP4 216.143.130.90. t=0 0. m=audio 17192 RTP/AVP 0 8 18 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U UA.IP.ADDRESS:63718 -> PROXY.IP.ADDRESS:5060 SIP/2.0 410 Gone. Via: SIP/2.0/UDP PROXY.IP.ADDRESS:5060;branch=z9hG4bK9159.cd2a95b4.0. Via: SIP/2.0/UDP PSTN.GW.IP:5090;branch=z9hG4bK39c17f01;rport=5090. From: <sip:13154476314@our.domain.com>;tag=as7bbe322e. To: "User One" <sip:1101XXXXXXX@our.domain.com>;tag=202d9942. Contact: <sip:1101XXXXXXX@UA.IP.ADDRESS:63718>. Call-ID: 6b3f44351f5fab945ac212a2141df93a(a)UA.IP.ADDRESS. CSeq: 102 INVITE. Content-Length: 0. First off, I thought a 410 was supposed to be unauthorised (what is 'Gone' ?). Secondly... is the UA requesting our server to authorise its reinvite, or is something else going on here I clearly don't follow? N.