<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
</head>
<body>
<p>if your going to do faxing at some point, I would test it now.
Its easier to play with the fax commands on the cube not worrying
about breaking things later.</p>
<p>Geesh, I could take all this info and probably build one heck of
a doc....<br>
</p>
<p><br>
</p>
<div class="moz-cite-prefix">On 2/11/22 10:34 AM, Matthew Huff
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:00eef99e161a410b8a7ddeea97eec2c5@ox.com">
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<meta name="Generator" content="Microsoft Word 15 (filtered
medium)">
<style>@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}@font-face
{font-family:Consolas;
panose-1:2 11 6 9 2 2 4 3 2 4;}p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri",sans-serif;}a:link, span.MsoHyperlink
{mso-style-priority:99;
color:#0563C1;
text-decoration:underline;}a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:#954F72;
text-decoration:underline;}pre
{mso-style-priority:99;
mso-style-link:"HTML Preformatted Char";
margin:0in;
margin-bottom:.0001pt;
font-size:10.0pt;
font-family:"Courier New";}p.msonormal0, li.msonormal0, div.msonormal0
{mso-style-name:msonormal;
mso-margin-top-alt:auto;
margin-right:0in;
mso-margin-bottom-alt:auto;
margin-left:0in;
font-size:11.0pt;
font-family:"Calibri",sans-serif;}span.HTMLPreformattedChar
{mso-style-name:"HTML Preformatted Char";
mso-style-priority:99;
mso-style-link:"HTML Preformatted";
font-family:Consolas;}span.EmailStyle21
{mso-style-type:personal;
font-family:"Calibri",sans-serif;
color:windowtext;}span.EmailStyle22
{mso-style-type:personal;
font-family:"Calibri",sans-serif;
color:windowtext;}span.EmailStyle23
{mso-style-type:personal-reply;
font-family:"Calibri",sans-serif;
color:windowtext;}.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}div.WordSection1
{page:WordSection1;}</style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
<div class="WordSection1">
<p class="MsoNormal">Thanks.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Our new SIP voice gateway is separate and
not in production so I have plenty of freedom to play.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">We have copper based FAX lines, not going
over our PRI currently. This is something we are looking into
though after this conversion is done.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<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" class="moz-txt-link-freetext">mhuff@ox.com</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">www.ox.com</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>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<div style="border:none;border-top:solid #E1E1E1
1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b>From:</b> Kent Roberts
<a class="moz-txt-link-rfc2396E" href="mailto:dvxkid@gmail.com"><dvxkid@gmail.com></a> <br>
<b>Sent:</b> Friday, February 11, 2022 12:14 PM<br>
<b>To:</b> Matthew Huff <a class="moz-txt-link-rfc2396E" href="mailto:mhuff@ox.com"><mhuff@ox.com></a>;
<a class="moz-txt-link-abbreviated" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] SIP to iTSP best
practices<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p>Oh yeah.. one more thing...<o:p></o:p></p>
<p>Test faxing!!!! a fax test is a min of 10 pages, inbound
call and out.... don't just do a page and say your good.
Check T38 if your using it... if you have to fail back because
of T38 non-compliant, is G711 working? Does your faxing
software do/support switchback to 711 if T38 doesn't setup. <o:p></o:p></p>
<p>If you have a fax machine on a ATA or whater, test to it as
well.<o:p></o:p></p>
<p><o:p> </o:p></p>
<p>Isn't fax dead yet? :) good luck with your go live.<o:p></o:p></p>
<p><o:p> </o:p></p>
<div>
<p class="MsoNormal">On 2/11/22 8:52 AM, Matthew Huff wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">Thanks for the recommendations. I have a
lot to dig into. Question about the video disable. We have
no video hardware, so think it would be good to disable it
before we go live. What’s the best way to disable it
globally?<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Is it <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Voice service voip<o:p></o:p></p>
<p class="MsoNormal"> Sip<o:p></o:p></p>
<p class="MsoNormal"> Audio forced<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>
<div>
<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</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:4.0pt;font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-AU"> </span><o:p></o:p></p>
<p class="MsoNormal"><i><span
style="font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-GB">Office: 914-460-4039</span></i><o:p></o:p></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" class="moz-txt-link-freetext">mhuff@ox.com</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">www.ox.com</a></span></i><o:p></o:p></p>
<p class="MsoNormal"><b><span
style="font-size:7.5pt;font-family:"Arial",sans-serif;color:gray">...........................................................................................................................................</span></b><o:p></o:p></p>
</div>
<p class="MsoNormal"> <o:p></o:p></p>
<div>
<div style="border:none;border-top:solid #E1E1E1
1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b>From:</b> Kent Roberts <a
href="mailto:dvxkid@gmail.com" moz-do-not-send="true">
<dvxkid@gmail.com></a> <br>
<b>Sent:</b> Thursday, February 10, 2022 6:14 PM<br>
<b>To:</b> Matthew Huff <a href="mailto:mhuff@ox.com"
moz-do-not-send="true"><mhuff@ox.com></a>; <a
href="mailto:cisco-voip@puck.nether.net"
moz-do-not-send="true" class="moz-txt-link-freetext">
cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] SIP to iTSP best
practices<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<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.<o:p></o:p></p>
<p>Recommendations:<o:p></o:p></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.<o:p></o:p></p>
<p>Use dedicated dial peers setup exactly for each itsp SBC
link for in and one for out.<o:p></o:p></p>
<p>Use something like the "voice class uri trunk(x) sip" or
equivalent to bind to the dial peers for each SBC.<o:p></o:p></p>
<p> This will help if you have to add additional
carriers, or say acquire a company, or need to do special
routing...<o:p></o:p></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)<o:p></o:p></p>
<div>
<p class="MsoNormal">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.
<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">define your voice class codecs on the
dialpeers... don't just assume it will take the default,
or work as you want without it.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">NEVER user debug ccsip all<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> debug CCSIP messages is safer,
and unless your cube is peeked, it won't add to much
cpu.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">make sure your CPU never exceeds 80%
at the max possible peek of routing.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Check how the calls work with MOH.
Inbound and out. make sure 2 way audio remains after
the on hold event..<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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)<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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)
<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> Some may allow calls to process
not caring what the number is.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> Some may allow you to provide a
alternate billing number.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> Some will just 603 decline the
call if the ANI isn't in your number poll assigned to
you.
<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> with a 603 the cube will try
the next dial peer so you can add a header to re-write
this with your number.....<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.)<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.
<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">define the FAX-relay, rats and
protocols for T38<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">ask your carrier how they handle
QOS. some don't since the trunk to them might be
dedicated.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Setup the event manager applet. have
it email you on syslog patterns for dialpeer status.
Then you will know if the link goes down.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">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.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Hope at least one thing here was
helpful.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">On 2/10/22 9:09 AM, Matthew Huff
wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<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</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:4.0pt;font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-AU"> </span><o:p></o:p></p>
<p class="MsoNormal"><i><span
style="font-family:"Arial",sans-serif;color:#1F497D"
lang="EN-GB">Office: 914-460-4039</span></i><o:p></o:p></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"
class="moz-txt-link-freetext">mhuff@ox.com</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">www.ox.com</a></span></i><o:p></o:p></p>
<p class="MsoNormal"><b><span
style="font-size:7.5pt;font-family:"Arial",sans-serif;color:gray">...........................................................................................................................................</span></b><o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>cisco-voip mailing list<o:p></o:p></pre>
<pre><a href="mailto:cisco-voip@puck.nether.net" moz-do-not-send="true" class="moz-txt-link-freetext">cisco-voip@puck.nether.net</a><o:p></o:p></pre>
<pre><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" moz-do-not-send="true" class="moz-txt-link-freetext">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></pre>
</blockquote>
</blockquote>
</blockquote>
</div>
</blockquote>
</body>
</html>