{cid_type=none,sip_h_Privacy=id}
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/2.0Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKSaXt6Bva8UD5KMax-Forwards: 70From: "Anonymous" <sip:9727289377@1.2.3.4;transport=udp>;tag=QjjNSt2m3g0rBTo: <sip:9735551234@5.6.7.8>Call-ID: 9b32aa4a-eaa3-122d-9bbb-000bdb94aab9CSeq: 131718385 INVITEContact: <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 -0400Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFYSupported: timer, precondition, path, replacesAllow-Events: talk, hold, referPrivacy: noneContent-Type: application/sdpContent-Disposition: sessionContent-Length: 291X-FS-Support: update_displayP-Asserted-Identity: "Anonymous" <sip:+19727289377@1.2.3.4>v=0o=FreeSWITCH 1275645559 1275645560 IN IP4 1.2.3.4s=FreeSWITCHc=IN IP4 1.2.3.4t=0 0m=audio 27884 RTP/AVP 0 8 3 101 13a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:13 CN/8000a=ptime:20Am I doing something incorrectly?
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