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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Hi,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<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...<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Thanks
for reading!!<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Neil<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
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