<div dir="ltr">Zahid,<div><br></div><div>You'd have to receive the initial INVITE, place it on hold somehow (not the SIP RFC sense of "hold", but just "in timeout", "on ice", "go stand in the corner until I tell you to leave", etc), wait for the INFO message, parse the information elements you want, recover the initial INVITE, update the fields you need (RPID/PAI, From, etc) and continue on about your business. I'm not sure how I would handle the putting-it-on-hold part, or the subsequent recovery. A proxy's job is generally to relay messages, and in this case, you want it to store a message for a period, then recover it and continue processing. That's not something a proxy normally does. So, that's a tricky one.</div><div><br></div><div><br></div><div>- Jeff</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Dec 18, 2015 at 10:12 AM, Zahid Mehmood <span dir="ltr"><<a href="mailto:zm23@columbia.edu" target="_blank">zm23@columbia.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Jeff,<div> Thanks for your response. From what I understand from Cisco documentation, gateway seems to act that way when using H323 but, sadly, not for SIP. I will push Cisco about this.</div><div><br></div><div>Assuming the worst, Is there any thing I can try on the proxy side to get the desired results?</div><div><br></div><div>Regards,</div><div>--</div><div>Zahid</div><div><br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Dec 18, 2015 at 9:44 AM, Jeff Pyle <span dir="ltr"><<a href="mailto:jeff.pyle@fidelityvoice.com" target="_blank">jeff.pyle@fidelityvoice.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">In Adtran TA900 series gateways (very Cisco-like) I'm able to configure the PRI interface to wait for the FACILITY message before sending the initial INVITE. When the INVITE does leave the gateway towards the proxy, it has full caller name information. Perhaps something like this is available on the Cisco. I hope so, because if not, you're going to have a difficult time integrating the INFO message.<div><br></div><div><br></div><div>- Jeff</div><div><br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Thu, Dec 17, 2015 at 2:53 PM, Zahid Mehmood <span dir="ltr"><<a href="mailto:zm23@columbia.edu" target="_blank">zm23@columbia.edu</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi,<div> I am having trouble figuring out how to process the calling-name coming from the PRI. In my setup, PRI is connected to a Cisco media gateway which sends traffic to the proxy servers. Calling name is not coming in the ISDN setup message. It is actually provided in a separate facility message [1].</div><div><br></div><div>Cisco gateway processes this secondary messages and generates a INFO message. Polycom phone sends the 200 ok message but there is no change in the visible caller id.</div><div><br></div><div><div>Does anyone have a working example or suggestion of how this is supposed to work? </div></div><div><br></div><div>Invite:</div><div><br></div><div><div>U 2015/12/17 14:20:31.215540 <a href="http://10.10.1.1:50975" target="_blank">10.10.1.1:50975</a> -> <a href="http://10.10.2.2:5060" target="_blank">10.10.2.2:5060</a></div><div>INVITE <a href="http://sip:10301@10.10.2.2:5060" target="_blank">sip:10301@10.10.2.2:5060</a> SIP/2.0.</div><div>Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3A2284.</div><div>Remote-Party-ID: "1112223333" <<a href="mailto:sip%3A1112223333@10.10.1.1" target="_blank">sip:1112223333@10.10.1.1</a>>;party=calling;screen=yes;privacy=off.</div><div>From: "1112223333" <<a href="mailto:sip%3A1112223333@10.10.1.1" target="_blank">sip:1112223333@10.10.1.1</a>>;tag=5745CCC-1C72.</div><div>To: <<a href="mailto:sip%3A10301@10.10.2.2" target="_blank">sip:10301@10.10.2.2</a>>.</div><div>Date: Thu, 17 Dec 2015 19:20:31 GMT.</div><div>Call-ID: <a href="mailto:12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1" target="_blank">12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1</a>.</div><div>Supported: 100rel,timer,resource-priority,replaces,sdp-anat.</div><div>Min-SE: 1800.</div><div>Cisco-Guid: 0311776101-2754220517-2148597785-1445067520.</div><div>User-Agent: Cisco-SIPGateway/IOS-12.x.</div><div>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.</div><div>CSeq: 101 INVITE.</div><div>Max-Forwards: 70.</div><div>Timestamp: 1450380031.</div><div>Contact: <<a href="http://sip:1112223333@10.10.1.1:5060" target="_blank">sip:1112223333@10.10.1.1:5060</a>>.</div><div>Expires: 180.</div><div>Allow-Events: telephone-event.</div><div>Content-Type: application/sdp.</div><div>Content-Disposition: session;handling=required.</div><div>Content-Length: 279.</div><div>.</div><div>v=0.</div><div>o=CiscoSystemsSIP-GW-UserAgent 3918 6190 IN IP4 10.10.1.1.</div><div>s=SIP Call.</div><div>c=IN IP4 10.10.1.1.</div><div>t=0 0.</div><div>m=audio 18854 RTP/AVP 0 18 101.</div><div>c=IN IP4 10.10.1.1.</div><div>a=rtpmap:0 PCMU/8000.</div><div>a=rtpmap:18 G729/8000.</div><div>a=fmtp:18 annexb=no.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div></div><div><br></div><div>Invite messages:</div><div><br></div><div><div>U 2015/12/17 14:20:31.546310 <a href="http://10.10.1.1:50975" target="_blank">10.10.1.1:50975</a> -> <a href="http://10.10.2.2:5060" target="_blank">10.10.2.2:5060</a></div><div>INFO <a href="http://sip:10301@10.219.136.69:5060" target="_blank">sip:10301@10.219.136.69:5060</a> SIP/2.0.</div><div>Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3C1EC0.</div><div>From: "1112223333" <<a href="mailto:sip%3A1112223333@10.10.1.1" target="_blank">sip:1112223333@10.10.1.1</a>>;tag=5745CCC-1C72.</div><div>To: <<a href="mailto:sip%3A10301@10.10.2.2" target="_blank">sip:10301@10.10.2.2</a>>;tag=1768D8EC-CCDB1323.</div><div>Date: Thu, 17 Dec 2015 19:20:31 GMT.</div><div>Call-ID: <a href="mailto:12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1" target="_blank">12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1</a>.</div><div>User-Agent: Cisco-SIPGateway/IOS-12.x.</div><div>Max-Forwards: 70.</div><div>Route: <sip:10.10.2.2;lr=on;ftag=5745CCC-1C72>.</div><div>Timestamp: 1450380031.</div><div>CSeq: 103 INFO.</div><div>Contact: <<a href="http://sip:1112223333@10.10.1.1:5060" target="_blank">sip:1112223333@10.10.1.1:5060</a>>.</div><div>Remote-Party-ID: "WIRELESS CALLER" <<a href="mailto:sip%3A1112223333@10.10.1.1" target="_blank">sip:1112223333@10.10.1.1</a>>;party=calling;screen=no;privacy=off.</div><div>Content-Length: 0.</div><div>.</div></div><div><br></div><div><br></div><div>Best Regards,</div><span><font color="#888888"><div><br></div><div>-- </div><div>Zahid</div><div><br></div><div>[1] <a href="http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-isdn.html#GUID-53D5C9AB-AAC4-4178-8158-0DAEFB5BC33E" target="_blank">http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-isdn.html#GUID-53D5C9AB-AAC4-4178-8158-0DAEFB5BC33E</a> (figure 2 is close to what we are seeing)</div></font></span></div>
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