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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Hi Guys,<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'>Dragging this one up again unfortunately!!<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'>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. <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'>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.<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'>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.<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'>Somewhere, this is failing and I’ve no idea where to look
at this point so any help is appreciated.<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>
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<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"'> Ryan Ratliff [mailto:rratliff@cisco.com] <br>
<b>Sent:</b> 13 May 2010 21:28<br>
<b>To:</b> O'Brien, Neil<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] SIP Trunk on CallManager Gateway Router<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<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>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>Try creating a dummy pots dial-peer and see if that works.<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>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<div>
<p class=MsoNormal>On May 13, 2010, at 4:20 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'>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><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'>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><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'>Am I missing something?</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'> </span><o:p></o:p></p>
</div>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm;
border-width:initial;border-color:initial'>
<div>
<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"'>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">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>
<p class=MsoNormal> <o:p></o:p></p>
</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>
<p class=MsoNormal> <o:p></o:p></p>
</div>
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<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>
</div>
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<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
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<div>
<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";
color:black'>-Ryan</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=MsoNormal>On May 13, 2010, at 3:32 PM, O'Brien, Neil wrote:<o:p></o:p></p>
</div>
</div>
<div>
<p class=MsoNormal><br>
<br>
<br>
<o:p></o:p></p>
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<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Hi,</span><o:p></o:p></p>
</div>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;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><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;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><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
</div>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;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><o:p></o:p></p>
</div>
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<div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
</div>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;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><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;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><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
</div>
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<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Thanks
for reading!!</span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </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"'>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"'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p>
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<p class=MsoNormal><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">cisco-voip@puck.nether.net</a><br>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a></span><o:p></o:p></p>
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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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