<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=Windows-1252">
<style type="text/css" style="display:none;"> P {margin-top:0;margin-bottom:0;} </style>
</head>
<body dir="ltr">
<div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);">
<div style="font-family: Calibri, Helvetica, sans-serif, serif, EmojiFont; font-size: 12pt">
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, serif, EmojiFont; font-size: 12pt">
<br>
</div>
<div style="font-family: Calibri, Helvetica, sans-serif, serif, EmojiFont; font-size: 12pt">
<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, serif, EmojiFont; font-size: 12pt">
<i><span style="font-size:8pt; line-height:normal">{ .. other config .. }</span></i></div>
<div style="font-family: Calibri, Helvetica, sans-serif, serif, EmojiFont; font-size: 12pt">
<i><span style="font-size:8pt; line-height:normal"><br>
</span></i></div>
<div style="font-family: Calibri, Helvetica, sans-serif, serif, EmojiFont; font-size: 12pt">
<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, serif, EmojiFont; font-size: 12pt">
<i><span style="font-size:8pt; line-height:normal">!</span></i></div>
<br>
</div>
<div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);">
<br>
</div>
<div id="signature">
<div></div>
<div id="divtagdefaultwrapper" style="font-size:12pt; color:#000000; background-color:#FFFFFF; font-family:Calibri,Arial,Helvetica,sans-serif">
Thanks,</div>
<div style="font-size:12pt; color:#000000; background-color:#FFFFFF; font-family:Calibri,Arial,Helvetica,sans-serif">
<br>
</div>
<div style="font-size:12pt; color:#000000; background-color:#FFFFFF; font-family:Calibri,Arial,Helvetica,sans-serif">
Ryan</div>
</div>
<div style="font-family:Calibri,Helvetica,sans-serif; font-size:12pt; color:rgb(0,0,0)">
<br>
</div>
<hr tabindex="-1" style="display:inline-block; width:98%">
<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>From:</b> Anthony Holloway <avholloway+cisco-voip@gmail.com><br>
<b>Sent:</b> Thursday, December 20, 2018 12:46 PM<br>
<b>To:</b> NateCCIE<br>
<b>Cc:</b> Ryan Huff; Erik Anderson; cisco-voip voyp list<br>
<b>Subject:</b> Re: [cisco-voip] SIP Fail over</font>
<div> </div>
</div>
<div>
<div dir="ltr">Agreed, but it would be terrible if they stated they don't support it, you bother them with OPTIONS, and then they black list you. Just be careful and get written approval, is all I'm saying.</div>
<br>
<div class="x_gmail_quote">
<div dir="ltr">On Thu, Dec 20, 2018 at 11:44 AM NateCCIE <<a href="mailto:nateccie@gmail.com">nateccie@gmail.com</a>> wrote:<br>
</div>
<blockquote class="x_gmail_quote" style="margin:0px 0px 0px 0.8ex; border-left:1px solid rgb(204,204,204); padding-left:1ex">
<div lang="EN-US">
<div class="x_gmail-m_-6682341490580999285WordSection1">
<p class="x_MsoNormal">When you say level3 does not support options ping, do you mean they won’t ping you for failover, or they don’t allow you to send it to them? Only two messages and the lack of any response will busy the endpoint, anything else if good
enough for CUBE.<u></u><u></u></p>
<p class="x_MsoNormal"><u></u> <u></u></p>
<p class="x_MsoNormal"><img id="x_gmail-m_-6682341490580999285Picture_x0020_1" style="width:2.9916in; height:1.9416in" width="287" height="186" data-outlook-trace="F:1|T:1" src="cid:167ccb998e34cff311"><u></u><u></u></p>
<p class="x_MsoNormal"><u></u> <u></u></p>
<div>
<div style="border-right:none; border-bottom:none; border-left:none; border-top:1pt solid rgb(225,225,225); padding:3pt 0in 0in">
<p class="x_MsoNormal"><b>From:</b> cisco-voip <<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>>
<b>On Behalf Of </b>Ryan Huff<br>
<b>Sent:</b> Thursday, December 20, 2018 10:35 AM<br>
<b>To:</b> Erik Anderson <<a href="mailto:erik.anderson.85@gmail.com" target="_blank">erik.anderson.85@gmail.com</a>>; cisco-voip voyp list <<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Subject:</b> Re: [cisco-voip] SIP Fail over<u></u><u></u></p>
</div>
</div>
<p class="x_MsoNormal"><u></u> <u></u></p>
<div>
<p class="x_MsoNormal"><span style="font-size:12pt; color:black">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...<u></u><u></u></span></p>
</div>
<div>
<p class="x_MsoNormal"><span style="font-size:12pt; color:black"><u></u> <u></u></span></p>
</div>
<div id="x_gmail-m_-6682341490580999285signature">
<div id="x_gmail-m_-6682341490580999285divtagdefaultwrapper">
<p class="x_MsoNormal" style="background:white"><span style="font-size:12pt; color:black">Thanks,<u></u><u></u></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12pt; color:black"><u></u> <u></u></span></p>
</div>
<div>
<p class="x_MsoNormal" style="background:white"><span style="font-size:12pt; color:black">- Ryan<u></u><u></u></span></p>
</div>
</div>
<div class="x_MsoNormal" style="text-align:center" align="center">
<hr width="98%" size="3" align="center">
</div>
<div id="x_gmail-m_-6682341490580999285divRplyFwdMsg">
<p class="x_MsoNormal"><b><span style="color:black">From:</span></b><span style="color:black"> 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</span> <u></u><u></u></p>
<div>
<p class="x_MsoNormal"> <u></u><u></u></p>
</div>
</div>
<div>
<div>
<p class="x_MsoNormal">Morning Folks, <u></u><u></u></p>
<div>
<p class="x_MsoNormal"><br clear="all">
<u></u><u></u></p>
<div>
<p>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.<u></u><u></u></p>
<p> <u></u><u></u></p>
<p>Architecture Notes:<u></u><u></u></p>
<p> <u></u><u></u></p>
<p>4 Locations with 1 CUBE Each<u></u><u></u></p>
<p>4 CUCM SIP Trunks with each connecting to one of the 4 CUBEs<u></u><u></u></p>
<p>4 CUCM Route Groups with Various CUBE/SIP Trunks assigned a Distribution Algorithm of
<i>Top Down</i><u></u><u></u></p>
<p>Each CUBE has 2 SIP Peers<u></u><u></u></p>
<p>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<u></u><u></u></p>
<p>Level 3 does not support SIP Options Ping<u></u><u></u></p>
<p>CUCM Trunks have SIP Options Ping enabled<u></u><u></u></p>
<p> <u></u><u></u></p>
<p>Call Flows:<u></u><u></u></p>
<p> <u></u><u></u></p>
<p style="margin-left:27pt">Working Flow:<u></u><u></u></p>
<p> <u></u><u></u></p>
<p style="margin-left:27pt">Phone ----> SLRG ----> Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call Completes<u></u><u></u></p>
<p style="margin-left:3.75in"> <u></u><u></u></p>
<p style="margin-left:3.75in"> <u></u><u></u></p>
<p style="margin-left:27pt">CUBE Failure:<u></u><u></u></p>
<p> <u></u><u></u></p>
<p style="margin-left:27pt">Phone ----> SLRG ----><u></u><u></u></p>
<p style="margin-left:81pt"> Route Group Member #1 ----> CUBE SIP TRUNK --X--> CUBE (CUCM Cant Reach CUBE)<u></u><u></u></p>
<p style="margin-left:135pt"> <u></u><u></u></p>
<p style="margin-left:81pt"> CUCM Routes Call to Next Route Group Member<u></u><u></u></p>
<p style="margin-left:135pt"> <u></u><u></u></p>
<p style="margin-left:27pt"> Route Group Member #2 ----> CUBE SIP TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call Completes<u></u><u></u></p>
<p style="margin-left:27pt"> <u></u><u></u></p>
<p style="margin-left:27pt">Level 3 Transport Failure/SIP Server Failure:<u></u><u></u></p>
<p> <u></u><u></u></p>
<p style="margin-left:27pt">Phone ----> SLRG ----><u></u><u></u></p>
<p style="margin-left:81pt"> Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE --X--> Level 3 Transport (CUBE Cant Reach Level 3 SIP Server)<u></u><u></u></p>
<p style="margin-left:135pt"> <u></u><u></u></p>
<p style="margin-left:81pt"> CUCM Thinks Call Connects since the CUBE accepts the call, Phone gets dead air, never tries the next RG Member<u></u><u></u></p>
<p style="margin-left:135pt"> <u></u><u></u></p>
<p> <u></u><u></u></p>
<p>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.
<u></u><u></u></p>
<p> <u></u><u></u></p>
<p>If any has any other better ideas please let me know.<u></u><u></u></p>
</div>
<p class="x_MsoNormal">-- <u></u><u></u></p>
<div>
<p class="x_MsoNormal">Erik Anderson<u></u><u></u></p>
</div>
</div>
<div>
<p class="x_MsoNormal">Telecom Manager<u></u><u></u></p>
</div>
<div>
<p class="x_MsoNormal">Some Random Corp.<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<a href="https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7C%7C98ff5afcff2d4fc72c9208d666a32825%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C636809248267412528&sdata=ewI%2FZZIuvSBncKe2cGId8eXr3WIrCLkJx4XL4C%2FkuPk%3D&reserved=0" originalsrc="https://puck.nether.net/mailman/listinfo/cisco-voip" shash="rTP7Otf+odoOrYv5qWgzkg3vayp7bJ9gb60iP8dpXLEL7B+9fGbcV7J+Lf1AYX857l8fuggUXb/9GoPIOAOyIwClP8iijEmZ50c2i3vwrVvEb+QtjltwQRxu8CjlXhTdph5IWvE5IkKkKxEhkWFBScSMvMapS49AKLrp/UhIhcg=" rel="noreferrer" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
</blockquote>
</div>
</div>
</body>
</html>