[Freeswitch-users] Privacy: id / Inbound

[Liam Farr]

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 at 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 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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