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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Thanks Ryan,<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'>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></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'><br>
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>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<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"'> Ryan Ratliff [mailto:rratliff@cisco.com] <br>
<b>Sent:</b> 28 May 2010 14:21<br>
<b>To:</b> O'Brien, Neil<br>
<b>Cc:</b> Frank Arrasmith; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk on CallManager Gateway Router<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>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></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>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></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<div>
<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";
color:black'>-Ryan<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<div>
<div>
<p class=MsoNormal>On May 27, 2010, at 5:01 PM, O'Brien, Neil wrote:<o:p></o:p></p>
</div>
<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
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<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?</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Thanks,</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Neil</span><o:p></o:p></p>
</div>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span class=apple-converted-space><span
lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> </span></span><span
lang=EN-US style='font-size:10.0pt;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></p>
</div>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<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>
<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>
<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>Hi Guys,</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>Dragging this
one up again unfortunately!!</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><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>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><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>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><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>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><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>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>Thanks,</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>Neil</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt'>From:</span></b><span
class=apple-converted-space><span lang=EN-US style='font-size:10.0pt'> </span></span><span
lang=EN-US style='font-size:10.0pt'>Ryan Ratliff [mailto:<a
href="mailto:rratliff@cisco.com" target="_blank">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></p>
</div>
<div>
<div>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt'><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">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></p>
</div>
</div>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>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>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
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</div>
<div>
<div>
<p class=MsoNormal>Try creating a dummy pots dial-peer and see if that works.<o:p></o:p></p>
</div>
</div>
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<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=MsoNormal><span style='font-size:13.5pt;color:black'>-Ryan</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
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<div>
<p class=MsoNormal>On May 13, 2010, at 4:20 PM, O'Brien, Neil wrote:<o:p></o:p></p>
</div>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'> <o:p></o:p></p>
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<div>
<p class=MsoNormal><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>
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<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
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<div>
<p class=MsoNormal><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>
<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
</div>
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<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>Am I missing
something?</span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
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</div>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
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<p class=MsoNormal><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>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>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>
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<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=MsoNormal>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>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:13.5pt;color:black'>-Ryan</span><o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<div>
<p class=MsoNormal>On May 13, 2010, at 3:32 PM, O'Brien, Neil wrote:<o:p></o:p></p>
</div>
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<p class=MsoNormal style='margin-bottom:12.0pt'><br>
<br>
<br>
<o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'>Hi,</span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<div>
<p class=MsoNormal><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>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><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>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'>Thanks for reading!!</span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'>Neil</span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:13.5pt'>_______________________________________________<br>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><br>
_______________________________________________<br>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif"'>_______________________________________________<br>
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<p class=MsoNormal><o:p> </o:p></p>
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