<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">Here is the config I used long time ago for CVP, and really the same for CUCM. Still prefer the dial-peer sip options for timeout</div><div class=""><br class=""></div><div class=""><br class=""></div><div class="">ip host _sip._tcp.cvp.ucce.local srv 1 50 5060 nedcvpb.ucce.local</div><div class="">ip host _sip._tcp.cvp.ucce.local srv 2 50 5060 acccvpa.ucce.local</div><div class=""><br class=""></div><div class="">ip host _sip._udp.cvp.ucce.local srv 1 50 5060 nedcvpb.ucce.local</div><div class="">ip host _sip._udp.cvp.ucce.local srv 2 50 5060 acccvpa.ucce.local</div><div class=""><br class=""></div><div class=""><div class=""><br class=""></div><div class="">ip host acccvpa.ucce.local 10.80.19.11</div><div class="">ip host nedcvpb.ucce.local 10.42.14.21</div></div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><div class="">dial-peer voice 222222 voip</div><div class=""> session protocol sipv2</div><div class=""> session target sip-server</div><div class=""> incoming called-number 4444222T</div><div class=""><br class=""></div></div><div class=""><div class="">sip-ua </div><div class=""> retry invite 2</div><div class=""> retry bye 2</div><div class=""> retry cancel 2</div><div class=""> timers expires 60000</div><div class=""> sip-server dns:cvp.ucce.local</div><div class=""> reason-header override</div></div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><br class=""></div><div><blockquote type="cite" class=""><div class="">On Nov 18, 2019, at 10:00 PM, Kent Roberts <<a href="mailto:kent@fredf.org" class="">kent@fredf.org</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html; charset=utf-8" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Oh I know what your trying to do, and I have had issues with it as well. <div class=""><br class=""></div><div class=""> I have had better luck with with just using several dial-peers, and preference them, and use the voice-class sip options keepalive </div><div class=""><br class=""></div><div class="">Then when you look at the dial-peer status you can see them out of service, and if you have SNMP enabled, you can see the status when integrated by things like Orion…..</div><div class=""><br class=""><div class=""><br class=""><blockquote type="cite" class=""><div class="">On Nov 18, 2019, at 9:56 PM, Jonathan Charles <<a href="mailto:jonvoip@gmail.com" class="">jonvoip@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">I pasted the wrong part of the script (to manually change it)...<div class=""><br class=""></div><div class="">Here is the actual config:</div><div class=""><br class=""></div><div class=""><br class=""><br class="">voice class server-group 1<br class=""> ipv4 172.31.120.43<br class=""> ipv4 172.31.125.43 preference 2<br class=""> description Verizon SIP<br class="">!<br class=""></div><div class=""><br class=""></div><div class="">Jonathan</div></div><br class=""><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, Nov 18, 2019 at 10:22 PM Anthony Holloway <<a href="mailto:avholloway%2Bcisco-voip@gmail.com" class="">avholloway+cisco-voip@gmail.com</a>> wrote:<br class=""></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr" class="">First off, I'm wondering why it says "no ipv4" in front of your two addresses. That might be your problem right there.<div class=""><br class=""></div><div class="">Secondly, I'd recommend putting an explicit preference on your entries, it's just better for everyone, and you don't get a credit back from Cisco for saving on a few ascii characters by implicitly using the default. Plus, if the default is 0, which it is, then your next preference should be technically 1. But then having nothing and 1 seems silly, because if pref 1 is actually pref 2, then well, might as well call them pref nothing and pref 8. I digress.</div><div class=""><br class=""></div><div class="">You might not have failed over, because you might not have provided the system with the correction conditions to failover...E.g., you didn't wait long enough.</div><div class=""><br class=""></div><div class="">No seriously, by default SIP failover occurs after 30 seconds. Unless, did you lower the retry count under sip-ua? Or did you enable SIP options? If you enabled SIP options, have your confirmed that it's turned on correctly?</div><div class=""><br class=""></div><div class="">Can you share the output of the following commands:</div><div class=""><br class=""></div><div class="">show run | section sip-ua|sip.options-keepalive</div><div class=""><br class=""></div><div class="">show dial-peer voice summary</div><div class=""><br class=""></div><div class="">Feel free to redact what you need to, in terms of IPs or usernames/passwords. I am only looking for the features and settings for retries and keepalives.</div><div class=""><br class=""></div></div><br class=""><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, Nov 18, 2019 at 9:26 PM Jonathan Charles <<a href="mailto:jonvoip@gmail.com" target="_blank" class="">jonvoip@gmail.com</a>> wrote:<br class=""></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr" class="">Using session server groups on outbound dial-peers and it does not appear to be failing over:<div class=""><br class=""></div><div class=""><br class="">voice class server-group 1<br class=""> no ipv4 172.31.125.43
preference 2
<br class=""> no ipv4 172.31.120.43 <br class=""> description Verizon SIP<br class="">!<br class=""></div><div class=""><br class=""></div><div class="">We had the 172.31.20.43 go down (no response to invites) and we did NOT failover to the second (.125.43)...</div><div class=""><br class=""></div><div class="">What is needed to force a failover to the next configured SBC?</div><div class=""><br class=""></div><div class=""><br class=""></div><div class="">Jonathan</div></div>
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