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</head><body lang="EN-US" link="#0563C1" vlink="#954F72" style="zoom: 0%;"><div dir="auto">One more that is missing ... DTMF. Check it both ways . You may be suprised how different ITSP's are handling those and tricks you need to get them working (even getting PVDM just for it if you don't use transcoders or creating DTMF map).<br><br></div>
<div dir="auto">For video disabled, we handle it (more than 90 countries) by region in CUCM to VGW where we have only Audio possible (G711Only). Of course there are other possibilities directly on VGW.</div>
<div class="gmail_quote" >W dniu 11 lut 2022, o 16:58, użytkownik Matthew Huff <<a href="mailto:mhuff@ox.com" target="_blank">mhuff@ox.com</a>> napisał:
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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<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?</p><p></p>
<p class="MsoNormal"></p><p> </p>
<p class="MsoNormal">Is it </p><p></p>
<p class="MsoNormal"></p><p> </p>
<p class="MsoNormal">Voice service voip</p><p></p>
<p class="MsoNormal"> Sip</p><p></p>
<p class="MsoNormal"> Audio forced</p><p></p>
<p class="MsoNormal"></p><p> </p>
<p class="MsoNormal">?</p><p></p>
<p class="MsoNormal"></p><p> </p>
<div>
<p class="MsoNormal"><b><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D">Matthew Huff</span></b><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D"> | Director of Technical Operations | OTA Management LLC</span></p><p></p><p></p>
<p class="MsoNormal"><span lang="EN-AU" style="font-size:4.0pt;font-family:"Arial",sans-serif;color:#1F497D"></span></p><p> </p><p></p>
<p class="MsoNormal"><i><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D">Office: 914-460-4039</span></i></p><p></p><p></p>
<p class="MsoNormal"><i><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D"><a href="mailto:mhuff@ox.com">mhuff@ox.com</a> | </span></i><i><span lang="EN-GB" style="font-family:"Arial",sans-serif"><a href="http://www.ox.com">www.ox.com</a><span style="color:#1F497D"></span></span></i></p><p></p><p></p>
<p class="MsoNormal"><b><span style="font-size:7.5pt;font-family:"Arial",sans-serif;color:gray">...........................................................................................................................................</span></b></p><p></p><p></p>
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<div style="border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b>From:</b> Kent Roberts <dvxkid@gmail.com> <br>
<b>Sent:</b> Thursday, February 10, 2022 6:14 PM<br>
<b>To:</b> Matthew Huff <mhuff@ox.com>; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] SIP to iTSP best practices</p><p></p>
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<p class="MsoNormal"></p><p> </p>
<p class="MsoNormal"><br>
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<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.</p><p></p>
<p>Recommendations:</p><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.</p><p></p>
<p>Use dedicated dial peers setup exactly for each itsp SBC link for in and one for out.</p><p></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></p>
<p> This will help if you have to add additional carriers, or say acquire a company, or need to do special routing...</p><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)</p><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.
</p><p></p>
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<div>
<p class="MsoNormal"></p><p> </p>
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<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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
</div>
<div>
<p class="MsoNormal">NEVER user debug ccsip all</p><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.</p><p></p>
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<div>
<p class="MsoNormal"></p><p> </p>
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<div>
<p class="MsoNormal">make sure your CPU never exceeds 80% at the max possible peek of routing.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<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..</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<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)</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<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)
</p><p></p>
</div>
<div>
<p class="MsoNormal"> Some may allow calls to process not caring what the number is.</p><p></p>
</div>
<div>
<p class="MsoNormal"> Some may allow you to provide a alternate billing number.</p><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.
</p><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.....</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.)</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<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.
</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
</div>
<div>
<p class="MsoNormal">define the FAX-relay, rats and protocols for T38</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><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.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<div>
<p class="MsoNormal">Hope at least one thing here was helpful.</p><p></p>
</div>
<div>
<p class="MsoNormal"></p><p> </p>
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<p class="MsoNormal"></p><p> </p>
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<p class="MsoNormal"></p><p> </p>
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<div>
<p class="MsoNormal">On 2/10/22 9:09 AM, Matthew Huff wrote:</p><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?</p><p></p>
<p class="MsoNormal"> </p><p></p>
<p class="MsoNormal"> </p><p></p>
<p class="MsoNormal"> </p><p></p>
<p class="MsoNormal"><b><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D">Matthew Huff</span></b><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D"> | Director of Technical Operations | OTA Management LLC</span></p><p></p>
<p class="MsoNormal"><span lang="EN-AU" style="font-size:4.0pt;font-family:"Arial",sans-serif;color:#1F497D"> </span></p><p></p>
<p class="MsoNormal"><i><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D">Office: 914-460-4039</span></i></p><p></p>
<p class="MsoNormal"><i><span lang="EN-GB" style="font-family:"Arial",sans-serif;color:#1F497D"><a href="mailto:mhuff@ox.com">mhuff@ox.com</a> | </span></i><i><span lang="EN-GB" style="font-family:"Arial",sans-serif"><a href="http://www.ox.com">www.ox.com</a></span></i></p><p></p>
<p class="MsoNormal"><b><span style="font-size:7.5pt;font-family:"Arial",sans-serif;color:gray">...........................................................................................................................................</span></b></p><p></p>
<p class="MsoNormal"> </p><p></p>
<p class="MsoNormal"><br>
<br>
</p><p></p>
<pre>_______________________________________________<p></p></pre>
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</blockquote>
</blockquote>
</div>
<pre class="blue"><hr><br>cisco-voip mailing list<br>cisco-voip@puck.nether.net<br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br></pre></blockquote></div></body></html>