I have now resolved this issue with; <condition field="privacy_hide_number" expression="true" continue="true" break="never"> <action application="set" data="sip_h_Privacy=id"/> <action application="set" data="effective_caller_id_number=Unknown"/> <action application="set" data="effective_caller_id_name=Unknown"/> </condition> On Fri, May 27, 2011 at 4:43 PM, Liam Farr <liam@intersys.co.nz> wrote:

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