[Freeswitch-users] Privacy: id / Inbound

[Liam Farr]

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 at 202.55.98.6 SIP/2.0

   Max-Forwards: 69

   Supported: 100rel

   To: <sip:92568888 at 202.55.98.6>

   From: <sip:225553389 at 23.51.15.101>;tag=3515458594-499967

   P-Asserted-Identity: <sip:225553389 at 23.51.15.101>

   Privacy: id

   Call-ID: 23213457-3515458594-499959 at 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 at 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 at 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 at 202.55.98.7>;tag=8eK40BpUtXSUB

   To: <sip:555 at 202.55.98.122:5080>

   Call-ID: a7bce49f-02ba-122f-a19a-8903425aa839

   CSeq: 12898770 INVITE

   Contact: <sip:mod_sofia at 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 at 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
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