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<div class=Section1>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Hi Frank – it’s not a CUBE router. Forgive my ignorance but
should it be?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Neil<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Frank Arrasmith
[mailto:frank.arrasmith@gmail.com] <br>
<b>Sent:</b> 27 May 2010 18:40<br>
<b>To:</b> O'Brien, Neil<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk on CallManager Gateway Router<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>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>
<p class=MsoNormal>On Thu, May 27, 2010 at 4:15 AM, O'Brien, Neil <<a
href="mailto:nobrien@datapac.com">nobrien@datapac.com</a>> wrote:<o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>Hi Guys,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>Dragging this one up again
unfortunately!!</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>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></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>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></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>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></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>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></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>Thanks,</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>Neil</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0cm 0cm 0cm;
border-color:-moz-use-text-color -moz-use-text-color'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
lang=EN-US style='font-size:10.0pt'>From:</span></b><span lang=EN-US
style='font-size:10.0pt'> Ryan Ratliff [mailto:<a
href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>] <br>
<b>Sent:</b> 13 May 2010 21:28<o:p></o:p></span></p>
<div>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt'><br>
<b>To:</b> O'Brien, Neil<br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk on CallManager Gateway Router<o:p></o:p></span></p>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Try
creating a dummy pots dial-peer and see if that works.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:13.5pt;color:black'>-Ryan</span><o:p></o:p></p>
</div>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On
May 13, 2010, at 4:20 PM, O'Brien, Neil wrote:<o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><o:p> </o:p></p>
<div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>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></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>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></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'>Am I missing something?</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0cm 0cm 0cm;
border-color:-moz-use-text-color -moz-use-text-color'>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
lang=EN-US style='font-size:10.0pt'>From:</span></b><span lang=EN-US
style='font-size:10.0pt'> Ryan Ratliff [mailto:<a
href="mailto:rratliff@cisco.com" target="_blank">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">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk on CallManager Gateway Router</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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></p>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:13.5pt;color:black'>-Ryan</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On
May 13, 2010, at 3:32 PM, O'Brien, Neil wrote:<o:p></o:p></p>
</div>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><br>
<br>
<o:p></o:p></p>
</div>
<div>
<div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>Hi,</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'> </span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>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></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'> </span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>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></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'> </span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>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></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'> </span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>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></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'> </span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>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></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'> </span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt'>Thanks for reading!!</span><o:p></o:p></p>
</div>
</div>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
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