<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">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.<div><br></div><div>Try creating a dummy pots dial-peer and see if that works.</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>
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<br><div><div>On May 13, 2010, at 4:20 PM, 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, 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.<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); ">The router only seems to register with the provider when the phones register with the router (ie. Go into srst)<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); ">Am I missing something?<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>13 May 2010 21:12<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" 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; ">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 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; ">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 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 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 13, 2010, at 3:32 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; ">Hi,</span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">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><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">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><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">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><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">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><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">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><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">Thanks for reading!!</span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; ">Neil</span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><span style="font-size: 10pt; font-family: Verdana, sans-serif; "> </span><span style="font-size: 11pt; font-family: Calibri, sans-serif; "><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; "><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></div></span></blockquote></div><br></div></body></html>