<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Not necessarily, but it will introduce a bit of complexity as the router has to do the h.323 to sip internetworking.<div><br></div><div>In this case you should see CUCM send an H.225 Connect message to the router, which should generate a SIP 200 OK out to the provider, but possibly not until the router has completed H.245 media negotiations with CUCM.</div><div><br></div><div>If what I'm saying doesn't make sense it's time to engage TAC.</div><div><br><div>
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div>-Ryan</div></span>
</div>
<br><div><div>On May 28, 2010, at 9:38 AM, O'Brien, Neil wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div lang="EN-IE" link="blue" vlink="purple" style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div class="Section1"><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); ">Thanks Ryan,<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); ">But the CUCM isn’t talking SIP at all though, the gateway is still a H323 gateway. Is this my problem??<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); ">Thanks,<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "><br>Neil<o:p></o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; padding-top: 3pt; padding-right: 0cm; padding-bottom: 0cm; padding-left: 0cm; "><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span lang="EN-US" style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span lang="EN-US" style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span>Ryan Ratliff [mailto:rratliff@cisco.com]<span class="Apple-converted-space"> </span><br><b>Sent:</b><span class="Apple-converted-space"> </span>28 May 2010 14:21<br><b>To:</b><span class="Apple-converted-space"> </span>O'Brien, Neil<br><b>Cc:</b><span class="Apple-converted-space"> </span>Frank Arrasmith;<span class="Apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class="Apple-converted-space"> </span>Re: [cisco-voip] SIP Trunk on CallManager Gateway Router<o:p></o:p></span></div></div></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">It may be mostly semantics but at the point your router is doing IP to IP it's acting as a CUBE. Traditional voice gateways are IP to POTS.<o:p></o:p></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">I'd disagree that your signaling is good however. If the phone answers, and that is not communicated back out the SIP trunk then something is not right in the signaling.<o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">You need to look at the SIP messaging on the CUCM and at the router, just to make sure everything CUCM sends is passed on.<o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 13.5pt; font-family: Helvetica, sans-serif; color: black; ">-Ryan<o:p></o:p></span></div></div></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">On May 27, 2010, at 5:01 PM, O'Brien, Neil wrote:<o:p></o:p></div></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><br><br><o:p></o:p></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); ">Hi Frank – it’s not a CUBE router. Forgive my ignorance but should it be?</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); ">Thanks,</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); ">Neil</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; padding-top: 3pt; padding-right: 0cm; padding-bottom: 0cm; padding-left: 0cm; border-width: initial; border-color: initial; "><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span lang="EN-US" style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span class="apple-converted-space"><span lang="EN-US" style="font-size: 10pt; font-family: Tahoma, sans-serif; "> </span></span><span lang="EN-US" style="font-size: 10pt; font-family: Tahoma, sans-serif; ">Frank Arrasmith [mailto:frank.arrasmith@gmail.com]<span class="apple-converted-space"> </span><br><b>Sent:</b><span class="apple-converted-space"> </span>27 May 2010 18:40<br><b>To:</b><span class="apple-converted-space"> </span>O'Brien, Neil<br><b>Subject:</b><span class="apple-converted-space"> </span>Re: [cisco-voip] SIP Trunk on CallManager Gateway Router</span><o:p></o:p></div></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div><p class="MsoNormal" style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 12pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">This is just a shot, but it sounds like the signaling is good, so maybe something with the media? Is your CUBE set for flow-through, or flow-around? When I set mine up, I had 2 problems, I messed with the media settings, and changed to flow-around, which my equipment didn't support. It should be flow-through, which I think is default. The other problem I had was with binding the media to an interface. I either had problems with routing to the interface, or bound the media to the wrong interface, I can't remember off the top of my head. Just a couple more areas to look at...Please post back if you find a solution, as SIP/CUBE issues seem to pop up more and more these days.<br><br>--Frank<o:p></o:p></p><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">On Thu, May 27, 2010 at 4:15 AM, O'Brien, Neil <<a href="mailto:nobrien@datapac.com" style="color: blue; text-decoration: underline; ">nobrien@datapac.com</a>> wrote:<o:p></o:p></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Hi Guys,</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Dragging this one up again unfortunately!!</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">So when the phones register back to CCM, I have a dummy pots dial-peer (as Ryan suggested) created so the SIP trunk remains registered.</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">So at this point, all phones are registered to CCM. I call in on the SIP trunk, the phone rings, when I answer nothing happens and the caller continues to hear ringback, then both disconnect.</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Logically, the way I see it is that the SIP call comes in on the IOS gateway and hits the incoming SIP dial-peer, it gets bumped over to the CCM via another dial-peer and CCM rings the phone in question. This is all signalling. What should then happen is CCM connects the phone and the SIP call and then drops out of the loop so the phone is talking directly with it’s local gateway that terminates the sip trunk.</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Somewhere, this is failing and I’ve no idea where to look at this point so any help is appreciated.</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Thanks,</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Neil</span><o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; padding-top: 3pt; padding-right: 0cm; padding-bottom: 0cm; padding-left: 0cm; border-width: initial; border-color: initial; "><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span lang="EN-US" style="font-size: 10pt; ">From:</span></b><span class="apple-converted-space"><span lang="EN-US" style="font-size: 10pt; "> </span></span><span lang="EN-US" style="font-size: 10pt; ">Ryan Ratliff [mailto:<a href="mailto:rratliff@cisco.com" target="_blank" style="color: blue; text-decoration: underline; ">rratliff@cisco.com</a>]<span class="apple-converted-space"> </span><br><b>Sent:</b><span class="apple-converted-space"> </span>13 May 2010 21:28</span><o:p></o:p></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span lang="EN-US" style="font-size: 10pt; "><br><b>To:</b><span class="apple-converted-space"> </span>O'Brien, Neil<br><b>Cc:</b><span class="apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net" target="_blank" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class="apple-converted-space"> </span>Re: [cisco-voip] SIP Trunk on CallManager Gateway Router</span><o:p></o:p></div></div></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">Not my area of expertise, but a colleague pointed out that in order for the router to register with the sip provider it needs a pots dial-peer. This happens automatically when the phones register via SRST.<o:p></o:p></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">Try creating a dummy pots dial-peer and see if that works.<o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 13.5pt; color: black; ">-Ryan</span><o:p></o:p></div></div></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">On May 13, 2010, at 4:20 PM, O'Brien, Neil wrote:<o:p></o:p></div></div></div><p class="MsoNormal" style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 12pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></p><div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Thanks ryan, but when the phones register with CM (ie. Come out of SRST) incoming SIP calls don’t even hit the router and the provider sees it as not registered.</span><o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">The router only seems to register with the provider when the phones register with the router (ie. Go into srst)</span><o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); ">Am I missing something?</span><o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; color: rgb(31, 73, 125); "> </span><o:p></o:p></div></div></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; padding-top: 3pt; padding-right: 0cm; padding-bottom: 0cm; padding-left: 0cm; border-width: initial; border-color: initial; "><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span lang="EN-US" style="font-size: 10pt; ">From:</span></b><span lang="EN-US" style="font-size: 10pt; "> Ryan Ratliff [mailto:<a href="mailto:rratliff@cisco.com" target="_blank" style="color: blue; text-decoration: underline; ">rratliff@cisco.com</a>] <br><b>Sent:</b> 13 May 2010 21:12<br><b>To:</b> O'Brien, Neil<br><b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] SIP Trunk on CallManager Gateway Router</span><o:p></o:p></div></div></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">In CCM 4.x nothing should be talking SIP to CCM. You should be just fine doing H.323 to the gateway and SIP out from there. This is referred to as a CUBE, or IP-IP gateway setup.<o:p></o:p></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">If the calls aren't even hitting the router as H.323 ('debug h225 asn1' doesn't show anything) then you need to look at the CCM config. If the call is reaching the router but not going out SIP then check to make sure you have enabled the appropriate 'allow connection' commands under 'voice service voip'.<o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div><div><div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 13.5pt; color: black; ">-Ryan</span><o:p></o:p></div></div></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div><div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; ">On May 13, 2010, at 3:32 PM, O'Brien, Neil wrote:<o:p></o:p></div></div></div></div><div><p class="MsoNormal" style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 12pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><br><br><br><o:p></o:p></p></div><div><div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">Hi,</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">A customer of mine has Call Manager 4.x and a few remote sites. One of the sites was connected via an MPLS network and had a PSTN connection on their local gateway router. They moved premises and ditched the MPLS connection and instead of getting a new PSTN line, they went down the SIP road and got a SIP trunk to an internet provider. All the phones continued to work via SRST to the local router and the router was configured with the SIP trunk details as if it were a Call Manager Express setup.</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">The remote site now has a VPN back to the HQ and the phones are now registering back with Call Manager. Their local gateway router is still configured as that site’s Call Manager H323 gateway and I thought that once Call Manager pushed the calls to the gateway the call would continue to go out over the SIP trunk.</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">Unfortunately, the SIP trunk only seems to register with the provider when the phones register in SRST on the router. When the phones register back to call manager, the router will not register with the SIP provider. Outbound calls will still go out over the SIP but incoming calls will not work. I’m told by the provider that if the router isn’t registered with the SIP provider, outbound calls will still go over it but inbound calls will not.</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">It’s not a dial peer routing issue as when I debug the dial peers while the phones are registered to Call Manager and make an inbound call I don’t see anything hit the router. When the register in SRST, I see the SIP calls come in and match the relevant dial peers.</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">I’m really at a loss as to why this would be, I don’t know very well how SIP works with Call Manager. I’d really appreciate some pointers on this...</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">Thanks for reading!!</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; ">Neil</span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div><div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 10pt; "> </span><o:p></o:p></div></div></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 13.5pt; ">_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a></span><o:p></o:p></div></div></div></div></div><div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div></div></div></div></div></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div></div></div></div></div><p class="MsoNormal" style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 12pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></p></div><div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "> <o:p></o:p></div></div></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 13.5pt; font-family: Helvetica, sans-serif; ">_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></span></div></div></div><div style="margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div></div></div></span></blockquote></div><br></div></body></html>