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@vallimamod.org .
On 3 Feb 2017, at 15:24, Christoph Russow <russow@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@88.198.223.120:5060 SIP/2.0 Via: SIP/2.0/UDP 217.110.38.229:5060;branch=z9hG4bK0dB7a5469c2220f98ee From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 To: <sip:4980058xxxxx@88.198.223.120> Call-ID: 1863167222_96798667@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@217.110.38.229:5060> P-Asserted-Identity: <sip:499132xxxxx@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@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@88.198.223.120>;tag=eDe24j5Hp5cvm To: <sip:49151708xxxxx@80.84.31.98> Call-ID: 736df04f-64b0-1235-ad92-b8ac6f7db265 CSeq: 102726678 INVITE Contact: <sip:mod_sofia@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@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@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
-- ________________________________________________________________
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