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%3A9735551234@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 <sip%3A9727289377@1.2.3.4> ;transport=udp>;tag=QjjNSt2m3g0rB To: <sip:9735551234@5.6.7.8 <sip%3A9735551234@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<sip%3A%2B19727289377@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?