I tried setting the sip_h_Privacy via:

freeswitch.Session("{sip_cid_type=pid,origination_caller_id_name=" .. caller_id_name .. ",origination_caller_id_number=+1" .. cid .. ",sip_h_Privacy=id,ignore_early_media=true}sofia/gateway/mygateway/" .. call_this_number)

but in the SIP Capture Privacy is still none:

   INVITE sip:9735551234@5.6.7.8 SIP/2.0
   Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKSaXt6Bva8UD5K
   Max-Forwards: 70
   From: "Anonymous" <sip:9727289377@1.2.3.4;transport=udp>;tag=QjjNSt2m3g0rB
   To: <sip:9735551234@5.6.7.8>
   Call-ID: 9b32aa4a-eaa3-122d-9bbb-000bdb94aab9
   CSeq: 131718385 INVITE
   Contact: <sip:gw+verizon1@1.2.3.4:5080;transport=udp;gw=verizon1>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 291
   X-FS-Support: update_display
   P-Asserted-Identity: "Anonymous" <sip:+19727289377@1.2.3.4>

   v=0
   o=FreeSWITCH 1275645559 1275645560 IN IP4 1.2.3.4
   s=FreeSWITCH
   c=IN IP4 1.2.3.4
   t=0 0
   m=audio 27884 RTP/AVP 0 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20

Am I doing something incorrectly?