Problems with privacy settings on invite
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 -- ________________________________________________________________ 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 ________________________________________________________________
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
-- ________________________________________________________________
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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, it seems that one of our carriers sees the number in the FROM: field (which freeswitch pulled from inbound P-Asserted-Identity Header) and forwards them to the enduser. Inbound is: From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 P-Asserted-Identity: <sip:499132xxxxx@217.110.38.229:5060> Outbound is: From: "Anonymous" <sip:499132xxxxx@88.198.223.120>;tag=eDe24j5Hp5cvm P-Asserted-Identity: "Anonymous" <sip:499132xxxxx@88.198.223.120> Wanted on Outbound: From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 P-Asserted-Identity: <sip:499132xxxxx@217.110.38.229:5060> So i now want to tell freeswitch either to make the outbound call totaly anonymous (how?) and i set the P-Asserted-Identity manually. Or tell freeswitch to not pull the P-Asserted-Identity Contents into from field on the outbound call. Do you mean Setting the origination_caller_id_number or sip_from_user to anonymous would work here? is it the preferred way for privacy? Our Carriers don't require registering and so i did not see any advantage to create a profile for every carrier we start to use as it needs manual accessing the server and not only adding it into our database and let the scripts decide which one to use. Best Regards Christoph Am 03.02.2017 um 18:08 schrieb Vallimamod Abdullah:
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,
-- ________________________________________________________________ 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 ________________________________________________________________
Hi, You can manage the PID and privacy headers manually with something like this: bridge {origination_uuid=...,sip_cid_type=none,sip_h_P-Asserted-Identity=sip:499132xxxxx@88.198.223.120,sip_h_Privacy=id}sofia/... You can add "sip_from_uri=sip:Anonymous@Anonymous.invalid" if the from header is not as you want it. You don't need to create a new profile for every carrier, just the corresponding gateway. Even if there is no registration, you can monitor the gateways with sip options and do failover if needed. In your case, the gateway also allows you to define "caller-id-in-from" parameter which tells freeswitch to not use the pid as the outgoing from. But the same is achieved with the "sip_cid_type=none" setting above. Best Regards, -- Vallimamod Abdullah VOIP Consultant vma@vallimamod.org .
On 6 Feb 2017, at 10:50, Christoph Russow <russow@emtex.de> wrote:
Hi,
it seems that one of our carriers sees the number in the FROM: field (which freeswitch pulled from inbound P-Asserted-Identity Header) and forwards them to the enduser.
Inbound is: From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 P-Asserted-Identity: <sip:499132xxxxx@217.110.38.229:5060>
Outbound is: From: "Anonymous" <sip:499132xxxxx@88.198.223.120>;tag=eDe24j5Hp5cvm P-Asserted-Identity: "Anonymous" <sip:499132xxxxx@88.198.223.120>
Wanted on Outbound: From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 P-Asserted-Identity: <sip:499132xxxxx@217.110.38.229:5060>
So i now want to tell freeswitch either to make the outbound call totaly anonymous (how?) and i set the P-Asserted-Identity manually.
Or tell freeswitch to not pull the P-Asserted-Identity Contents into from field on the outbound call.
Do you mean Setting the origination_caller_id_number or sip_from_user to anonymous would work here? is it the preferred way for privacy?
Our Carriers don't require registering and so i did not see any advantage to create a profile for every carrier we start to use as it needs manual accessing the server and not only adding it into our database and let the scripts decide which one to use.
Best Regards Christoph
Am 03.02.2017 um 18:08 schrieb Vallimamod Abdullah:
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,
Hi, sorry for the delayed answer. thank you for you input i will try that. best regards christoph Am 06.02.2017 um 14:11 schrieb Vallimamod Abdullah:
Hi,
You can manage the PID and privacy headers manually with something like this:
bridge {origination_uuid=...,sip_cid_type=none,sip_h_P-Asserted-Identity=sip:499132xxxxx@88.198.223.120,sip_h_Privacy=id}sofia/...
You can add "sip_from_uri=sip:Anonymous@Anonymous.invalid" if the from header is not as you want it.
You don't need to create a new profile for every carrier, just the corresponding gateway. Even if there is no registration, you can monitor the gateways with sip options and do failover if needed. In your case, the gateway also allows you to define "caller-id-in-from" parameter which tells freeswitch to not use the pid as the outgoing from. But the same is achieved with the "sip_cid_type=none" setting above.
Best Regards,
-- ________________________________________________________________ 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 ________________________________________________________________
If you want your dialplan being dynamic, jst export sip_from_uri hdr from leg A to leg B. Instead of putting in your dialstring: "sip_from_uri=sip:Anonymous@Anonymous.invalid". Srigo
On 8 Feb 2017, at 12:13, Christoph Russow <russow@emtex.de> wrote:
Hi,
sorry for the delayed answer.
thank you for you input i will try that.
best regards christoph
Am 06.02.2017 um 14:11 schrieb Vallimamod Abdullah: Hi,
You can manage the PID and privacy headers manually with something like this:
bridge {origination_uuid=...,sip_cid_type=none,sip_h_P-Asserted-Identity=sip:499132xxxxx@88.198.223.120,sip_h_Privacy=id}sofia/...
You can add "sip_from_uri=sip:Anonymous@Anonymous.invalid" if the from header is not as you want it.
You don't need to create a new profile for every carrier, just the corresponding gateway. Even if there is no registration, you can monitor the gateways with sip options and do failover if needed. In your case, the gateway also allows you to define "caller-id-in-from" parameter which tells freeswitch to not use the pid as the outgoing from. But the same is achieved with the "sip_cid_type=none" setting above.
Best Regards,
-- ________________________________________________________________
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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Christoph Russow
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Srigo Kana
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Vallimamod Abdullah