[Freeswitch-users] Problems with privacy settings on invite

Fri Feb 3 20:08:27 MSK 2017

Hi,

You can try to set origination_caller_id_number or sip_from_user variables in your bridge dial-string.
Normally, with "Privacy:id", the caller-id should appear fully restricted to the callee. Is it not the case?

Is there any specific reason why you do not define a gateway for your provider and dial by ip instead? 
If you define a gateway, you can set the caller-id-in-from param to use the inbound callerid in the from field of the outgoing call.

Best Regards,
-- 
Vallimamod Abdullah
VOIP Consultant
vma at vallimamod.org
.

> On 3 Feb 2017, at 15:24, Christoph Russow <russow at emtex.de> wrote:
> 
> Hi All,
> 
> i recently got informed by a collegue that it seems im not setting
> privacy stuff correct on our outbound calls.
> 
> After investigating the mentioned call i see the following invite on the
> inbound leg (i xxxxxed out the last 5 digits of every phone number):
> 
> INVITE sip:4980058xxxxx at 88.198.223.120:5060 SIP/2.0
> Via: SIP/2.0/UDP 217.110.38.229:5060;branch=z9hG4bK0dB7a5469c2220f98ee
> From: "Anonymous" <sip:Anonymous at Anonymous.invalid>;tag=gK0d356f75
> To: <sip:4980058xxxxx at 88.198.223.120>
> Call-ID: 1863167222_96798667 at 217.110.38.229
> CSeq: 21390 INVITE
> Max-Forwards: 26
> Allow:
> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
> Accept: application/sdp, application/isup, application/dtmf,
> application/dtmf-relay,  multipart/mixed
> Contact: "Anonymous" <sip:Anonymous at 217.110.38.229:5060>
> P-Asserted-Identity: <sip:499132xxxxx at 217.110.38.229:5060>
> Privacy: id
> Supported: timer,100rel
> Session-Expires: 1800
> Min-SE: 90
> Content-Length:  262
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
> 
> v=0
> o=Sonus_UAC 29264 8032 IN IP4 217.110.38.229
> s=SIP Media Capabilities
> c=IN IP4 217.110.38.228
> t=0 0
> m=audio 19174 RTP/AVP 18 8 100
> a=rtpmap:18 G729/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:100 telephone-event/8000
> a=fmtp:100 0-15
> a=sendrecv
> a=ptime:20
> 
> and on the Outbound Leg i see freeswitch sending the following invite to
> our outbound carrier:
> 
> INVITE sip:49151708xxxxx at 80.84.31.98 SIP/2.0
> Via: SIP/2.0/UDP 88.198.223.120;rport;branch=z9hG4bKFyctt2ZcKvBaD
> Max-Forwards: 25
> From: "Anonymous" <sip:499132xxxxx at 88.198.223.120>;tag=eDe24j5Hp5cvm
> To: <sip:49151708xxxxx at 80.84.31.98>
> Call-ID: 736df04f-64b0-1235-ad92-b8ac6f7db265
> CSeq: 102726678 INVITE
> Contact: <sip:mod_sofia at 88.198.223.120:5060>
> User-Agent: Emtex Voice Switch v1.9
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Privacy: id
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 251
> X-FS-Support: update_display,send_info
> P-Asserted-Identity: "Anonymous" <sip:499132xxxxx at 88.198.223.120>
> 
> v=0
> o=FreeSWITCH 1486105285 1486105286 IN IP4 88.198.223.120
> s=FreeSWITCH
> c=IN IP4 88.198.223.120
> t=0 0
> m=audio 20200 RTP/AVP 8 0 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> 
> my system is starting the outbound leg with the following command:
> 
> bridge
> {origination_uuid=0043c244-7f46-4484-a4a1-fa8863f22321,sip_cid_type=pid,origination_privacy=hide_name:hide_number:screen}sofia/default/49151708xxxxx at 80.84.31.98
> 
> The questions now are:
> 
> Is there any way to prevent freeswitch from pulling the informations
> from the P-Asserted-Identity header into the From: header in the invite?
> 
> Am i doing something wrong while setting the privacy options (any option
> that cancel each other out)?
> 
> What is the preferred way of getting a fully anonymous invite (caller
> number nowhere mentioned) but correctly set P-Asserted-Identity with the
> caller number (required by german carriers by law) (like the inbound leg)?
> 
> If you need any additional informations from me just ask.
> 
> Best regards
> Christoph Russow
> 
> -- 
> ________________________________________________________________
> 
> EMTEX GmbH
> Christoph Russow
> Software Engineer
> 
> Bischof-Otto-Weg 9
> D-91086 Aurachtal
>                          Geschäftsführer:       Markus Enzinger
> Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
> Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
> ________________________________________________________________
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org