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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I am not sure about the SIP-to-SIP not working. This was working years ago, as I helped a very large medical firm begin to utilize SIP to telco. At the time, I was able to experiment with all types of internetworking:<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:Symbol;color:#1F497D'><span style='mso-list:Ignore'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP to SIP to CUCM<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:Symbol;color:#1F497D'><span style='mso-list:Ignore'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP to H323 to CUCM<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='font-size:11.0pt;font-family:Symbol;color:#1F497D'><span style='mso-list:Ignore'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP to H323 with Gatekeeper to CUCM<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Now as for SIP to H323 – I don’t believe I would recommend this even if it was supported due to the consumption of resources. I know that you can do H323 to SIP utilizing Cisco’s Session Management Edition.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Also, we have several Enterprise Contact Centers utilizing SIP to telco and SIP proxy – using CUBE. One item missing from CUBE is CPA (call progress analysis) – though this should be coming soon.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>With SIP, one has to understand the capabilities of the opposing party. I have seen issues from ptime, DTMF-relay, codec and offer type. As for re-invite, you want to make sure that if you are transferring the call around internally that you do not pass the re-invite back out to the telco –as they most likely will not know what to do with it. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>What is the use case of the re-invite?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Kind Regards - <o:p></o:p></span></p><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Bob Zanett<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Technical Services Architect<o:p></o:p></span></p></div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Jonathan Charles<br><b>Sent:</b> Thursday, February 09, 2012 10:46 AM<br><b>To:</b> Nick Matthews<br><b>Cc:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> Re: [cisco-voip] CUBE Issues with SIP Reinvite...<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><br><br>I am pretty sure SIP to SIP on a cube is not supported, or at least it doesn't work.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I have tried connecting a SIP trunk from CUCM to the CUBE (and a SIP trunk to the provider) and the calls always fail (I need to try it again and debug why...)<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>The solution here was the SIP Session Expires timer set to 8 hours, now we add that to our invite and they just don't prompt us again.... of course, you will need your provider to care that you sent it...<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Jonathan<o:p></o:p></p></div><div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Feb 7, 2012 at 4:23 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>H.323-SIP isn't recommended. I'm also not sure if SIP profiles<br>actually doing anything in H.323-SIP.<br><br>I would try this SIP-SIP to see if you have better luck.<br><br>-nick<o:p></o:p></p><div><div><p class=MsoNormal><br>On Tue, Feb 7, 2012 at 3:32 PM, Roger Wiklund <<a href="mailto:roger.wiklund@gmail.com">roger.wiklund@gmail.com</a>> wrote:<br>>> We have spiked the SIP Session Expires Timer to 7800 seconds, but that has<br>>> had no effect... the SIP provider says they are conforming to the SIP<br>>> standard and it is us that is configured incorrectly.<br>>><br>>> Any ideas?<br>><br>> I had re-invite problems a month ago with a SIP-SIP CUBE running<br>> 15.1.4M IOS. Downgrading to 15.1.2 or up to 15.2 fixed the problem.<br>> Looks like a bug.<br>><br>> /Roger<br>><o:p></o:p></p></div></div><p class=MsoNormal>> _______________________________________________<br>> cisco-voip mailing list<br>> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal><br><br><span style='color:white'>itevomcid</span> <o:p></o:p></p></div></body></html>