Hi, When I receive an inbound call with Privacy: id set in the SIP invite freeswitch isn’t stripping off the caller I’d when passing the call onto an internal extension? The inbound invite looks like this; recv 1091 bytes from udp/[23.51.15.101]:5060 at 04:14:28.246255: ------------------------------------------------------------------------ INVITE sip:92568888@202.55.98.6 SIP/2.0 Max-Forwards: 69 Supported: 100rel To: <sip:92568888@202.55.98.6> From: <sip:225553389@23.51.15.101>;tag=3515458594-499967 P-Asserted-Identity: <sip:225553389@23.51.15.101> Privacy: id Call-ID: 23213457-3515458594-499959@MSX-101.squiggly.net CSeq: 1 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK Via: SIP/2.0/UDP 23.51.15.101:5060 ;branch=z9hG4bKf91a7732fbc823dc2c3252d3f7ec83fc Contact: <sip:225553389@23.51.15.101:5060;tgrp=2Degrees_sst03> Expires: 180 Call-Info: <sip:23.51.15.101>;method="NOTIFY;Event=telephone-event;Duration=1000" Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 v=0 o=MSX-101 8787 8322 IN IP4 23.51.15.101 s=sip call c=IN IP4 23.51.15.103 t=0 0 m=audio 29708 RTP/AVP 8 0 18 4 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 With the internal invite looking like this; send 1083 bytes to udp/[202.55.98.122]:5080 at 04:14:28.260497: ------------------------------------------------------------------------ INVITE sip:555@202.55.98.122:5080 SIP/2.0 Via: SIP/2.0/UDP 202.55.98.7;rport;branch=z9hG4bK20Bv5jcFpj00c Max-Forwards: 67 From: "225553389" <sip:225553389@202.55.98.7>;tag=8eK40BpUtXSUB To: <sip:555@202.55.98.122:5080> Call-ID: a7bce49f-02ba-122f-a19a-8903425aa839 CSeq: 12898770 INVITE Contact: <sip:mod_sofia@202.55.98.7:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display P-Asserted-Identity: "225553389" <sip:225553389@202.55.98.7> v=0 o=FreeSWITCH 1306450788 1306450789 IN IP4 202.55.98.7 s=FreeSWITCH c=IN IP4 202.55.98.7 t=0 0 m=audio 18880 RTP/AVP 8 9 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 As you can see the caller id is passed onto the internal extension, where I really need to strip it off / replace it with restricted / anonymous. Is there a way to catch this as a variable in the inbound dial plan? Thanks Liam