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<blockquote type="cite"
cite="mid:13597221-c280-575d-274c-c563b7056a53@gmail.com">
<p>I was part of the team that starting a large scale sip
migration almost 10 years ago. Have moved thousand's of DID
since then. Run multiple gig circuits into the cube.</p>
<p>Recommendations:</p>
<p>on the link to your provider, use address outside of the route
able block for your company. (say you use 10.x.x.x then use
172.16 or 192.168) If you can, don't route the itsp
connections on your company network, go direct to the routers
supporting those links. (BGP peers I would guess depending on
carrier/build) If you can use a dedicated router, unless is a
small site.... This is important if you wind up doing any kind
of call recording, or if you have to enable debugs during the
day.</p>
<p>Use dedicated dial peers setup exactly for each itsp SBC link
for in and one for out.</p>
<p>Use something like the "voice class uri trunk(x) sip" or
equivalent to bind to the dial peers for each SBC.</p>
<p> This will help if you have to add additional carriers, or
say acquire a company, or need to do special routing...<br>
</p>
<p>use full E164 to and from the carrier, they may only want to do
10 digit in/out, but that is easy enough. (uri trunkx will help
here, as the inbound number will be at the cube, then you can
route to cucm with outbound dial peer)<br>
</p>
<div class="moz-cite-prefix">From your CUCM still send the 9 or 8
or whatever for outbound, then strip on match in the dialpeer to
Itsp. This will keep call looping etc. <br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">define your voice class codecs on the
dialpeers... don't just assume it will take the default, or work
as you want without it.<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">if the cube will never see VIDEO,
disable the options. The cube software likes to release bugs
that cause the cube to go south with video errors.<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Depending on your carrier, you may
need to force G729 or G711 first, even if its not your preferred
codec, have seen were the SBC will not negotiate a call, if the
codecs aren't in the order the carriers SBC wants.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">do not assume the carriers network
will normalize the calls. For instance, if the destination is
on the same carrier, its a direct ip route via the SBC. If that
end side can't accept say G729 (cheaper sbc) the call will just
fail.<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">NEVER user debug ccsip all</div>
<div class="moz-cite-prefix"> debug CCSIP messages is safer,
and unless your cube is peeked, it won't add to much cpu.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">make sure your CPU never exceeds 80%
at the max possible peek of routing.<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Check how the calls work with MOH.
Inbound and out. make sure 2 way audio remains after the on
hold event..</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Do you need to force early offer?
(yes sounds silly, but have run into issues where some phones
had no audio unless this was set)<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Ask your carrier, how they handle
TFNs outbound, if you pass the ANI from a 3rd party. (this is
all billing stuff to the TFN owner) <br>
</div>
<div class="moz-cite-prefix"> Some may allow calls to process
not caring what the number is.</div>
<div class="moz-cite-prefix"> Some may allow you to provide a
alternate billing number.<br>
</div>
<div class="moz-cite-prefix"> Some will just 603 decline the
call if the ANI isn't in your number poll assigned to you. <br>
</div>
<div class="moz-cite-prefix"> with a 603 the cube will try
the next dial peer so you can add a header to re-write this with
your number.....</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Diversion headers exist, however most
carriers pass them through to the destination, and IVRs or Voice
Mail systems on the far side will try to process that
information, and do unexpected things. (the party your calling
doesn't exist for example.)<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">define the default sip control/media
source interface, this will be your destination from cucm. The
URI trucks will define the sip control/media on the ITSP side.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">If you use firewalls any where in
your company, that will pass voip... Set the rtp-port range on
the cube match the smaller range of what your going to use.
(say the old days 16384-32767) don't assume the firewall will
pass all the UDP ports by default.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">speaking of firewalls, check, double
check, and triple check, then do your own research if you will
use them, when it comes to SIP inspection. Some FW's have
options that need to be tweeked and defined, for the SIP port.
(this may control anything from timeouts, which media ports
engage) This is especially true with expressway in the DMZ.
It might be safer to not use sip inspection and just pass the
port. But for some FWs this is not true. <br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">define the FAX-relay, rats and
protocols for T38</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">ask your carrier how they handle
QOS. some don't since the trunk to them might be dedicated.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">use option pings on the dial peers,
so if the SBC goes away that dialpeer disables. The sbc side
just has to respond, even if its an error saying what is this...
that will keep the peer up.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Setup the event manager applet. have
it email you on syslog patterns for dialpeer status. Then you
will know if the link goes down.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">if you can get a bug scrub on the
version of IOS, don't be determined to use the newest code.
newest is not always best.<br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">Hope at least one thing here was
helpful.</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix"><br>
</div>
<div class="moz-cite-prefix">On 2/10/22 9:09 AM, Matthew Huff
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:6e3a800d480346c9895bc2c5fced7b41@ox.com">
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<p class="MsoNormal">We are in the process of migrating for a
legacy PTSN voice gateway (PRI) to a new CUBE based SIP
connection to a iTSP connected via a private metro ethernet
(not Internet based). Does anyone have a good source for
recipes / dial-plans recommendations / best practices for
this?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b><span
style="font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-GB">Matthew Huff</span></b><span
style="font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-GB"> | Director of Technical Operations | OTA
Management LLC<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:4.0pt;font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-AU"><o:p> </o:p></span></p>
<p class="MsoNormal"><i><span
style="font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-GB">Office: 914-460-4039<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span
style="font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-GB"><a href="mailto:mhuff@ox.com"
moz-do-not-send="true"><span style="color:#0563C1">mhuff@ox.com</span></a>
| </span></i><i><span
style="font-family:"Arial",sans-serif"
lang="EN-GB"><a href="http://www.ox.com"
moz-do-not-send="true"><span style="color:#0563C1">www.ox.com</span></a><span
style="color:#1F497D"><o:p></o:p></span></span></i></p>
<p class="MsoNormal"><b><span
style="font-size:7.5pt;font-family:"Arial",sans-serif;color:gray">...........................................................................................................................................<o:p></o:p></span></b></p>
<p class="MsoNormal"><o:p> </o:p></p>
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