<div dir="ltr">In my scenario, I have an srv record setup with 4 ccm nodes, 1 and 2 are equal weight and preferred, 3 and 4 have equal weights but are higher weighted than the 1/2 pair. ccm1 and 2 are at a main data center, ccm 3 and 4 are at a remote DC. I want calls load balanced to 1 and 2, and in the case those are unavailable, load balance between 3 and 4. This works as expected and is a great alternative to having 4 different (unneeded) dial peers.<div><br></div><div>The problem came to light when I added keepalive to this DP. everything is fine and dandy when 1/2 are available, but in testing failover (acl on an uplink to block comm to 1/2), the keepalive took the DP out of service. Perhaps there are timers that could be shortened or extended to provide a different response, but I decided I didn't really need keepalive when using SRV in this manner.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jul 21, 2016 at 2:31 PM, NateCCIE <span dir="ltr"><<a href="mailto:nateccie@gmail.com" target="_blank">nateccie@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div>I have only used SRV destination a little bit, but my CVP friends use them extensively. I have never seen the option ping with SRV records act differently than I would have expected.</div><div><br></div><div>Do you have experiences where the whole dialpeer went down and other members of the SRV were still accessable?<br><br>Sent from my iPhone</div><div><div class="h5"><div><br>On Jul 21, 2016, at 9:11 AM, Nick Barnett <<a href="mailto:nicksbarnett@gmail.com" target="_blank">nicksbarnett@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr">That may work fine based on how the SRV and options pings are configured... but if you are counting on an SRV record to point to another CCM sub when the primary is down (etc...), options pings will likely take the whole DP down when a single host in the SRV goes unreachable.</div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 20, 2016 at 6:12 PM, Erick Bergquist <span dir="ltr"><<a href="mailto:erickbee@gmail.com" target="_blank">erickbee@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">You could also look at adding keep alive (options ping) to the dial peers and call manager sip trunk with options mentioned above.<div><br></div><div>Do you mind sharing the tcl script?</div><span><font color="#888888"><div><br></div></font></span><div><span><font color="#888888">Erick</font></span><div><div><span></span><br><br>On Wednesday, July 20, 2016, Pawlowski, Adam <<a href="mailto:ajp26@buffalo.edu" target="_blank">ajp26@buffalo.edu</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="#0563C1" vlink="#954F72">
<div>
<p class="MsoNormal"><span style="color:#1f497d">Nathan,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"> Thanks, this looks to be exactly what I’m looking for, this way I don’t convey the wrong message. It doesn’t seem like I can have more than one option other than hunt or don’t hunt, and it seems
to be proper to let the telephony provider handle it. Cool. Thanks again.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d">Adam<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p>
<div style="border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt">
<div>
<div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Nathan Richardson [mailto:<a>nrichardson@gci.com</a>]
<br>
<b>Sent:</b> Wednesday, July 20, 2016 12:34 PM<br>
<b>To:</b> Pawlowski, Adam; '<a>cisco-voip@puck.nether.net</a>'<br>
<b>Subject:</b> RE: Route Dial-Peer Based On Response<u></u><u></u></span></p>
</div>
</div>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Verdana","sans-serif"">One thing that may help is to configure the “voice hunt” settings. For example, you could put in “no voice hunt temp-fail” which would make the router stop routing if it
receives a cause code 41 from the CM so it would skip your TCL script in that scenario and should send that code back to your ITSP. It may even work to combine “no voice hunt all” with “voice hunt unassigned-number” or something like that.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Verdana","sans-serif""><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Verdana","sans-serif""><a href="http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_v2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1190281" target="_blank">http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_v2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1190281</a><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Verdana","sans-serif""><u></u> <u></u></span></p>
<div>
<p class="MsoNormal">-Nathan Richardson<u></u><u></u></p>
</div>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Verdana","sans-serif""><u></u> <u></u></span></p>
<div>
<div style="border:none;border-top:solid #e1e1e1 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b>From:</b> cisco-voip [<a>mailto:cisco-voip-bounces@puck.nether.net</a>]
<b>On Behalf Of </b>Pawlowski, Adam<br>
<b>Sent:</b> Wednesday, July 20, 2016 5:33 AM<br>
<b>To:</b> '<a>cisco-voip@puck.nether.net</a>' <<a>cisco-voip@puck.nether.net</a>><br>
<b>Subject:</b> [cisco-voip] Route Dial-Peer Based On Response<u></u><u></u></p>
</div>
</div>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:12.0pt;font-family:"Times New Roman","serif";color:red">[External Email]</span><span style="font-size:12.0pt;font-family:"Times New Roman","serif"">
<u></u><u></u></span></p>
<p class="MsoNormal">Hey all,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"> I’ve set up our CUBE routers to try and be a bit more slick, so I am making use of e164 pattern maps, dial peer groups, and DNS SRV lookups for redundancy/randomization. All that actually seems to be working rather well.
I have a requirement to make any inactive/unallocated number in my UCM play a custome intercept. I did this, at least for now, by setting up a secondary dial peer that matches with a higher preference than my UCM peer, and it plays an announcement with a TCL
script. <u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"> I’d like to set this up so that if the UCM peer is down, or if it receives some other code indicating a temporary failure, etc, I either would like to bypass this peer so the code goes back to the ITSP, or I can play a message
saying something about technical difficulties, etc. I’m not sure it’s possible to do this? The other way of doing this would be to have the UCM itself with a translation or something to roll to an audiotext mailbox, which is how we do this today, but it requires
either that we maintain translations for all numbers, or a generic one that will answer to all extensions queried at the system which I don’t want to do either.<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"> Any thoughts?<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Regards,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Adam Pawlowski<u></u><u></u></p>
<p class="MsoNormal">SUNY Buffalo NCS<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
</div>
</div>
</blockquote></div></div></div>
<br>_______________________________________________<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://puck.nether.net/mailman/listinfo/cisco-voip" rel="noreferrer" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
<br></blockquote></div><br></div>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>cisco-voip mailing list</span><br><span><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a></span><br><span><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a></span><br></div></blockquote></div></div></div></blockquote></div><br></div>