<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Does this help?<div><a href="http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_9397_ps5640_TSD_Products_Configuration_Guide_Chapter.html">http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_9397_ps5640_TSD_Products_Configuration_Guide_Chapter.html</a></div><div><br></div><div><br><div apple-content-edited="true">
<span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; border-spacing: 0px; ">-Ryan</span>
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<br><div><div>On Apr 30, 2013, at 11:43 AM, Robert Kulagowski <<a href="mailto:rkulagow@gmail.com">rkulagow@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div>On Fri, Apr 26, 2013 at 2:50 PM, Robert Kulagowski <<a href="mailto:rkulagow@gmail.com">rkulagow@gmail.com</a>> wrote:<br><blockquote type="cite">On Fri, Apr 26, 2013 at 11:26 AM, Andreas Sikkema <<a href="mailto:asikkema@unet.nl">asikkema@unet.nl</a>> wrote:<br><blockquote type="cite">Hi,<br><br><blockquote type="cite">Is there some combination of privacy settings that I can enable on a<br>dial peer so that<br>1- the calling number and name is sent to the provider, so that they<br>can associate a calling and called number (and the cost of that call)<br>2- the upstream SIP provider _doesn't_ forward that information<br>towards the called party? For some calls it's important that they get<br>to the destination as "Unknown"<br></blockquote><br>This depends on which relevant SIP headers your upstream provider<br>supports for CLID presentation.<br><br>Some use the Remote-Party-ID header, others can handle<br>P-Preferred_Identity and/or P-Asserted-Identity headers with the<br>associated Privacy header that does the real restricting bit.<br></blockquote><br>"... <a href="http://bandwidth.com">bandwidth.com</a> uses the FROM field to represent the Caller ID Name<br>and Number & call rating. If a Remote-Party ID field (RPID) is<br>included in the SIP INVITE message, the RPID will be used for caller<br>ID and for call rating. The from field and the RPID must be in a<br>10-digit format."<br><br>OK, so in CUCM I set the Calling Line ID and Name Presentation to "Restricted".<br><br>In CUBE, I configured "privacy id" under voice service voip along with<br>"privacy-policy passthru"<br><br>The call leg from CUCM is:<br><br>INVITE <a href="sip:1630xxxxxxx@204.xxx.xxx.xxx:5060">sip:1630xxxxxxx@204.xxx.xxx.xxx:5060</a> SIP/2.0<br>Date: Fri, 26 Apr 2013 19:33:42 GMT<br>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,<br>SUBSCRIBE, NOTIFY<br>From: "Anonymous"<br><<a href="sip:anonymous@anonymous.invalid">sip:anonymous@anonymous.invalid</a>>;tag=d465acaa-c4d5-4a79-b7e9-2f1a79e6b2a6-39519923<br>Allow-Events: presence<br>P-Asserted-Identity: "Robert Kulagowski" <<a href="sip:223928@10.255.29.20">sip:223928@10.255.29.20</a>><br>Supported: timer,resource-priority,replaces<br>Supported: X-cisco-srtp-fallback<br>Supported: Geolocation<br>Min-SE: 1800<br>Remote-Party-ID: "Robert Kulagowski"<br><<a href="sip:223928@10.255.29.20">sip:223928@10.255.29.20</a>>;party=calling;screen=yes;privacy=full<br>Content-Length: 0<br>User-Agent: Cisco-CUCM7.1<br>Privacy: id<br>To: <<a href="sip:81630xxxxxxx@204.xxx.xxx.xxx">sip:81630xxxxxxx@204.xxx.xxx.xxx</a>><br>Contact: <<a href="sip:223928@10.255.29.20:5060">sip:223928@10.255.29.20:5060</a>><br>Expires: 180<br>Call-ID: 32807480-17a1d696-5-141dff0a@10.255.29.20<br>Via: SIP/2.0/UDP 10.255.29.20:5060;branch=z9hG4bK765b52845<br>CSeq: 101 INVITE<br>Session-Expires: 1800<br>Max-Forwards: 70<br><br>And what was sent to the provider:<br><br>INVITE <a href="sip:+1630xxxxxxx@ot.bandwidth.com:5060">sip:+1630xxxxxxx@ot.bandwidth.com:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 204.xx.xxx.xxx:5060;branch=z9hG4bK914FF<br>From: "anonymous" <<a href="sip:anonymous@204.xx.xxx.xxx">sip:anonymous@204.xx.xxx.xxx</a>>;tag=67C96C50-1048<br>To: <<a href="sip:+1630xxxxxxx@ot.bandwidth.com">sip:+1630xxxxxxx@ot.bandwidth.com</a>><br>Date: Fri, 26 Apr 2013 19:33:42 GMT<br>Call-ID: 9AB9FF1-<a href="mailto:ADDF11E2-86BFDAE7-AA8C8F4C@204.xx.xxx.xxx">ADDF11E2-86BFDAE7-AA8C8F4C@204.xx.xxx.xxx</a><br>Supported: 100rel,timer,resource-priority,replaces,sdp-anat<br>Min-SE: 1800<br>Cisco-Guid: 0162082489-2917077474-2260327143-2861338444<br>User-Agent: Cisco-SIPGateway/IOS-12.x<br>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,<br>SUBSCRIBE, NOTIFY, INFO, REGISTER<br>CSeq: 101 INVITE<br>Timestamp: 1367004822<br>Contact: <<a href="sip:anonymous@204.xx.xxx.xxx:5060">sip:anonymous@204.xx.xxx.xxx:5060</a>><br>Expires: 180<br>Allow-Events: telephone-event<br>Max-Forwards: 69<br>Session-Expires: 1800<br>Content-Type: application/sdp<br>Content-Disposition: session;handling=required<br>Content-Length: 253<br><br>Is there some combination of settings that will allow me to send the<br>"223928" part to the provider so that it shows up in their billing<br>system, but not send _anything_ to the person being called? I don't<br>want the called party to see "223928" on their phone, because that's<br>weird.<br></blockquote><br>Any ideas on this? Is this even possible? I've been reading through<br>the CVOICE material and I may be missing something, or just not<br>understanding everything fully.<br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>https://puck.nether.net/mailman/listinfo/cisco-voip<br><br></div></div><br></div></body></html>