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<div>“<span style="background-color: rgba(255, 255, 255, 0);">CCM syncs to a local ntp master set at 2 just fine; yields the CCM pub at S3 and cluster subs at
<b><i>S4</i></b>...”</span><br>
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<div id="AppleMailSignature" dir="ltr">Sent from my iPhone</div>
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On Dec 20, 2018, at 18:12, Ryan Huff <<a href="mailto:ryanhuff@outlook.com">ryanhuff@outlook.com</a>> wrote:<br>
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<div dir="ltr">CCM syncs to a local ntp master set at 2 just fine; yields the CCM pub at S3 and cluster subs at S3...
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<div>Alternatively, if your router syncs to an S1, and CCM Pub subsequently sync’ed to said router, you’ll find the CCM pub happily at S2 and the cluster subs then at an S3. Works great actually.</div>
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<div id="AppleMailSignature" dir="ltr">Sent from my iPhone</div>
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On Dec 20, 2018, at 18:05, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
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<div dir="ltr">Nate,
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<div>Good point. Might be hard to finesse a fake stratum into the master command, without accidentally messing with the selection process. Stratum number isn't the only criteria, and the process seems to be pretty complex.</div>
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<div>Looks like we might not be able to have our cake and eat it too.</div>
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<div>You must be referring to the following sentence from the <a href="https://nam02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2Fcucm%2Fsrnd%2Fcollab12%2Fcollab12%2Fnetstruc.html%23pgfId-1497943&data=02%7C01%7C%7C5e8677a191d94d1bce6f08d666cfb1c5%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C636809439553495449&sdata=J42WNMq8Q8hA1KH28asEQpw2Qjt0rHjiXUJ61Ahhu9U%3D&reserved=0" originalsrc="https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/netstruc.html#pgfId-1497943" shash="F3FKhNI2OqBjjavUOUgBYeASqu3LEZXSTfTxo07p7l37V6oNqMtecXfZpHIwpDQhWQqPNwa6LiEy9wEbRlltFwblZ6hBsddGPdbekCegXsGYdv7Gz5d3nJ5YcJVnfMX0rQCEkFsVTb1lA3Yz6vaVXk+sNtg0fr71l3W1k4UXKEA=">CUCM
SRND</a>:</div>
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<div>"Cisco highly recommends configuring the publisher to point to a Stratum-1, Stratum-2, or Stratum-3 NTP server to ensure that the cluster time is synchronized with an external time source."</div>
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<div dir="ltr">On Thu, Dec 20, 2018 at 4:46 PM NateCCIE <<a href="mailto:nateccie@gmail.com">nateccie@gmail.com</a>> wrote:<br>
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<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="auto">I think the lowest cucm will use is a 3?<br>
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<div id="gmail-m_-3970485705865814158AppleMailSignature" dir="ltr">Sent from my iPhone</div>
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On Dec 20, 2018, at 3:35 PM, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
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<div>I have never seen that done before. I like it!</div>
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<div dir="ltr">Just be careful hard coding your stratum to a value of 2 all the time. Instead it should be a relative value, higher than your reference clock. Or if you do want a one-size-fits-all stratum,
<a href="https://nam02.safelinks.protection.outlook.com/?url=http%3A%2F%2Fsupport.ntp.org%2Fbin%2Fview%2FSupport%2FSelectingOffsiteNTPServers%23Section_5.3.5.&data=02%7C01%7C%7C5e8677a191d94d1bce6f08d666cfb1c5%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C636809439553495449&sdata=JEd%2BRZotzTeLqrXcEx6rEzCjH1SKJhQkTfLzrS%2BEEPc%3D&reserved=0" originalsrc="http://support.ntp.org/bin/view/Support/SelectingOffsiteNTPServers#Section_5.3.5." shash="WFLJyxua/XmkKHoqdpiy/XillN2x8UKPoaCg6ZP5uFkVLjnz2JfJwKXH9iEHmtf7mr9nsfMrPhqMlv2g/JTPjd5Cm6slIRiU2l2TN3cly+bQe2kadBmIZ2dhEwvAZaDAay34RoLmRHI/MzXe65UK/ADB5Mt8AZMalbkKXHeuQO4=" target="_blank">
14 maybe</a>?</div>
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<div>Thanks for sharing that tip Ryan!</div>
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<div dir="ltr">On Thu, Dec 20, 2018 at 3:52 PM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>> wrote:<br>
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<div dir="auto">I like ntp master 2 as a fallback, to allow synchronization with the local device clock in a DR/Outage scenario where I fail sync to the actual reference clock<br>
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Sent from my iPhone</div>
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On Dec 20, 2018, at 14:51, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
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<div dir="ltr">It's very interesting to me the kinds of things people take for granted and go a long time without ever being corrected, simply because the people who know these things, think it's common knowledge.
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<div>For example, I had a conversation with a senior collab person once, who didn't know that region bit rate settings were a ceiling, and that a lower bit rate could be negotiated. </div>
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<div>And as another example, Engineers who put <b>ntp master</b> on a router because they think this makes the router an NTP server.</div>
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<div>And as one last example, Engineers who use the ^ symbol at the beginning of a Dial Peer destination pattern, not knowing that destination patterns are left justified implicitly. Or alternatively, don't use the $ at the end, effectively creating a "begins
with" clause, when an "is exactly" clause is desired.</div>
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<div>Someone should start a thread titled: What is something you found out that you were wrong about for a long time?</div>
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<div dir="ltr">On Thu, Dec 20, 2018 at 1:14 PM Lelio Fulgenzi <<a href="mailto:lelio@uoguelph.ca" target="_blank">lelio@uoguelph.ca</a>> wrote:<br>
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I’ll be honest. I didn’t know there was a difference.
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<div>I’m guessing a SIP trunk to a third party app that is reported as down due to to sip option ping really is down and not some silly networking issue where an icmp ping was failing. </div>
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<div>This is good to know. </div>
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<div>And the last thing I will learn this year. ;)</div>
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<p class="MsoNormal" style="margin:0in 0in 0.0001pt"><span style="background-color:rgba(255,255,255,0)"><b>Lelio Fulgenzi, B.A.</b> | Senior Analyst<u></u><u></u></span></p>
<p class="MsoNormal" style="margin:0in 0in 0.0001pt"><span style="background-color:rgba(255,255,255,0)">Computing and Communications Services | University of Guelph<u></u><u></u></span></p>
<p class="MsoNormal" style="margin:0in 0in 0.0001pt"><span style="background-color:rgba(255,255,255,0)">Room 037 Animal Science & Nutrition Bldg | <a dir="ltr">50 Stone Rd E | Guelph, ON | N1G 2W1</a><u></u><u></u></span></p>
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<p class="MsoNormal" style="margin:0in 0in 0.0001pt"><font color="#000000"><span style="background-color:rgba(255,255,255,0)"><a href="https://nam02.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs&data=02%7C01%7C%7C5e8677a191d94d1bce6f08d666cfb1c5%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C636809439553495449&sdata=bRO8hJZou38mNuKyBtfDmkGP7bbak34a5%2FKkRw6Oinc%3D&reserved=0" originalsrc="http://www.uoguelph.ca/ccs" shash="xns8CN8svOnAGBln+HjEexTd5LwG3yjKkjrYE4BLyeykLleQVRnzSLna2LWncWjQQV3n1x1Whq5wl/ctiiHeND1dybTQio11cVgIahmgXgU1HmOMoFHyMo+eyq99DHBxtxaRo1kT9HFN1nMTAtKR6szdVsQuU5ZtpN6oypLjtOw=" target="_blank">www.uoguelph.ca/ccs</a> |
@UofGCCS on Instagram, Twitter and Facebook<u></u><u></u></span></font></p>
<p class="MsoNormal" style="margin:0in 0in 0.0001pt"><span style="background-color:rgba(255,255,255,0)"> </span></p>
<p class="MsoNormal" style="margin:0in 0in 0.0001pt"><span style="background-color:rgba(255,255,255,0)"><img border="0" width="187" height="100" id="gmail-m_-3970485705865814158gmail-m_-1826016670442056916gmail-m_-5632280411951793794Picture_x0020_1" alt="University of Guelph Cornerstone with Improve Life tagline" action="http://null" title="null" style="width: 1.9479in; height: 1.0416in;" src="cid:"></span></p>
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On Dec 20, 2018, at 1:01 PM, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<br>
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<div>Erik,</div>
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That's an interesting insight. It kind of sounds like you think ICMP Ping and SIP OPTIONS Ping are related, but they are completely different.
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<div>Just because you cannot ICMP Ping the SIP Peer at L3, doesn't mean you cannot OPTIONs them.</div>
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<div>Am I understanding your thought process correctly?</div>
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<div dir="ltr">On Thu, Dec 20, 2018 at 11:53 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>> wrote:<br>
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<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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<div id="gmail-m_-3970485705865814158gmail-m_-1826016670442056916gmail-m_-5632280411951793794gmail-m_-5130095315229186341divtagdefaultwrapper" style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Thanks,</div>
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<div style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Ryan Huff, CCDP, CCNP<br>
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<div style="font-size:12pt;color:rgb(0,0,0);background-color:rgb(255,255,255);font-family:Calibri,Arial,Helvetica,sans-serif">
Cisco Certified Network and Design Professional</div>
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<div id="gmail-m_-3970485705865814158gmail-m_-1826016670442056916gmail-m_-5632280411951793794gmail-m_-5130095315229186341divRplyFwdMsg" dir="ltr">
<font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>><br>
<b>Sent:</b> Thursday, December 20, 2018 12:46 PM<br>
<b>To:</b> Erik Anderson<br>
<b>Subject:</b> Re: [cisco-voip] SIP Fail over</font>
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<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
Not sure what kind of code you're working with but if its modern, you could try server groups. Here is a snippet from one of mine (using AT&T admitidly), sanitized for the NSA ...</div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
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<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">voice class server-group 100</span><span><br>
</span></i>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">12.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 1</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">12.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">12.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 3</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 12.x.x.x preference 1</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description PSTN SIGNALING PEERS</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">voice class server-group 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 10.x.x.x preference 3</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">10.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 1</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> ipv4 </span><span style="font-size:8pt;line-height:normal">10.x.x.x</span><span style="font-size:8pt;line-height:normal"> preference 2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description CUCM SIGNALING PEERS</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">voice class sip-options-keepalive 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description PSTN HEARTBEAT</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">voice class sip-options-keepalive 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description CCM HEARTBEAT</span><br>
</i></div>
<i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">{ .. other config .. }</span></i></div>
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<i><span style="font-size:8pt;line-height:normal"><br>
</span></i></div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">dial-peer voice 100 voip</span><span><br>
</span></i>
<div><i><span style="font-size:8pt;line-height:normal"> description INGRESS/EGRESS WITH PSTN</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> translation-profile outgoing PLUS1_STRIP</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> huntstop</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination-pattern A</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session protocol sipv2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session server-group 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination dpg 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> incoming uri via PSTN</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class codec 1 </span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip options-ping 60</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip profiles 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip options-keepalive profile 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind control source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind media source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> dtmf-relay rtp-nte sip-notify</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> no vad</span></i></div>
<div><i><span style="font-size:8pt;line-height:normal">!</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal">dial-peer voice 200 voip</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> description INGRESS/EGRESS WITH CUCM</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> translation-profile outgoing PLUS1_STRIP</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> huntstop</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination-pattern A</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session protocol sipv2</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> session server-group 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> destination dpg 100</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> incoming uri via CUCM</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class codec 1 </span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip profiles 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip options-keepalive profile 200</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind control source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> voice-class sip bind media source-interface XXXX</span><br>
</i></div>
<div><i><span style="font-size:8pt;line-height:normal"> dtmf-relay rtp-nte sip-notify</span><br>
</i></div>
<i><span style="font-size:8pt;line-height:normal"> no vad</span></i></div>
<div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)">
<i><span style="font-size:8pt;line-height:normal">!</span></i><br>
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Thanks,</div>
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Ryan Huff, CCDP, CCNP<br>
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Cisco Certified Network and Design Professional</div>
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<font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Erik Anderson <<a href="mailto:erik.anderson.85@gmail.com" target="_blank">erik.anderson.85@gmail.com</a>><br>
<b>Sent:</b> Thursday, December 20, 2018 12:37 PM<br>
<b>To:</b> Ryan Huff<br>
<b>Subject:</b> Re: [cisco-voip] SIP Fail over</font>
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<div dir="ltr">Ryan,
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<div>Level 3 does not support options ping. If i try to ping the call control IP it will always fail. There is a separate pingable address, but I didnt think i could configure the options ping to use any address other than the target. </div>
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<div dir="ltr">On Thu, Dec 20, 2018 at 11:34 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>> wrote:<br>
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Couldn't you just use voice class sip options/keepalives to mark when the ITSP is down, so CUCM marks the trunk out of service and fails to the next route group member immediately (ideally, your secondary CUBE)? Seems like thats a more natural way of doing
it versus using IP SLAs...<br>
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Thanks,</div>
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- Ryan<br>
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<font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> cisco-voip <<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>> on behalf of Erik Anderson <<a href="mailto:erik.anderson.85@gmail.com" target="_blank">erik.anderson.85@gmail.com</a>><br>
<b>Sent:</b> Thursday, December 20, 2018 12:03 PM<br>
<b>To:</b> cisco-voip voyp list<br>
<b>Subject:</b> [cisco-voip] SIP Fail over</font>
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<div dir="ltr">Morning Folks,
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<div>
<p style="margin:0in;font-family:Calibri;font-size:11pt">We have implemented a new SIP solution with Level 3 and found that we have outbound calling failover issues. When a CUBE loses its ability to talk to its Level 3 Peer, but can still talk to CUCM outbound
calls will still connect to the CUBE, but fail connecting to Level 3. In turn CUCM still thinks the call is connected since the CUCM SIP trunk remains up to the CUBE.</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Architecture Notes:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">4 Locations with 1 CUBE Each</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">4 CUCM SIP Trunks with each connecting to one of the 4 CUBEs</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">4 CUCM Route Groups with Various CUBE/SIP Trunks assigned a Distribution Algorithm of
<span style="font-style:italic">Top Down</span></p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Each CUBE has 2 SIP Peers</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Each CUBE can only talk to its respective SIP peer via its local Level 3 Transport to reduce call control latency by not allowing it to use the DMVPN backup network</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Level 3 does not support SIP Options Ping</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">CUCM Trunks have SIP Options Ping enabled</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">Call Flows:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Working Flow:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Phone ----> SLRG ----> Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call Completes</p>
<p style="margin:0in 0in 0in 3.75in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 3.75in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">CUBE Failure:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Phone ----> SLRG ----></p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> Route Group Member #1 ----> CUBE SIP TRUNK --X--> CUBE (CUCM Cant Reach CUBE)</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> CUCM Routes Call to Next Route Group Member</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt"> Route Group Member #2 ----> CUBE SIP TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call Completes</p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Level 3 Transport Failure/SIP Server Failure:</p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 0.375in;font-family:Calibri;font-size:11pt">Phone ----> SLRG ----></p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE --X--> Level 3 Transport (CUBE Cant Reach Level 3 SIP Server)</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in 0in 0in 1.125in;font-family:Calibri;font-size:11pt"> CUCM Thinks Call Connects since the CUBE accepts the call, Phone gets dead air, never tries the next RG Member</p>
<p style="margin:0in 0in 0in 1.875in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">My idea to fix this is to use an IPSLA to ping the pingable address on the Level 3 SIP Servers. If both address are unreachable then shutdown the CUCM Dial-Peers. This doesn’t sounds like the best way
of fixing it, but it should work. </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt"> </p>
<p style="margin:0in;font-family:Calibri;font-size:11pt">If any has any other better ideas please let me know.</p>
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-- <br>
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Erik Anderson</div>
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Telecom Manager</div>
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Some Random Corp.</div>
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-- <br>
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Erik Anderson</div>
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