[Serusers] Meaning of and proper response for a 401

[SIP]

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 at 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 at our.domain.com>;tag=as7bbe322e.
To: "User One" <sip:1101XXXXXXX at our.domain.com>;tag=202d9942.
Contact: <sip:13154476314 at PSTN.GW.IP:5090>.
Call-ID: 6b3f44351f5fab945ac212a2141df93a at 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 at our.domain.com>;tag=as7bbe322e.
To: "User One" <sip:1101XXXXXXX at our.domain.com>;tag=202d9942.
Contact: <sip:1101XXXXXXX at UA.IP.ADDRESS:63718>.
Call-ID: 6b3f44351f5fab945ac212a2141df93a at 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.