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