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<p><font size="-1">I think your best option is KeepAlived; on
keepalived configuration you declare a script name where you
execute:</font></p>
<p><font size="-1">/usr/local/bin/opensips-cli -x mi
clusterer_shtag_set_active vip/3</font></p>
<p><font size="-1">to switch VIP TAG from one server to other.</font></p>
<p><font size="-1">In this case BYE go to the right place.</font></p>
<p><font size="-1">If anyone want translate from spanish to english,
I have a complete tutorial for OpenSIPs 3.1</font></p>
<p><font size="-1">Regards<br>
</font></p>
<pre class="moz-signature" cols="72">---
I'm SoCIaL, MayBe</pre>
<div class="moz-cite-prefix">El 19/01/2021 a las 10:40 a. m., Kevin
Wormington escribió:<br>
</div>
<blockquote type="cite"
cite="mid:FAC36B73-76CF-479C-A6CE-F645C4679C95@missouri-telecom.com">
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<div class="">I’m not using RTPEngine…the upstream proxies are
handling all media, NAT traversal, etc. so the OpenSIPS
instances can always reach the endpoints. I’m using clusterer
module to share the user location and dialogs with different
active tags per node. There is zero loss of media on
switch-over and sometimes a little longer PDD for new calls
during switchover until the upstream proxies detect the instance
down. The only part I can’t seem to get to work is handling
the final BYE for calls that were on the failed node originally.
The re-invite ping will correct end them but would like to be
able to fix it completely…but maybe that is not currently
possible.</div>
<div class=""><br class="">
</div>
<br class="">
<div class="">
<div dir="auto" style="caret-color: rgb(0, 0, 0); color: rgb(0,
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after-white-space;" class="">
<div>Thanks,<br class="">
<br class="">
Kevin<br class="">
</div>
</div>
</div>
<div>
<blockquote type="cite" class="">
<div class="">On Jan 19, 2021, at 9:31 AM, Social Boh via
Users <<a href="mailto:users@lists.opensips.org" class=""
moz-do-not-send="true">users@lists.opensips.org</a>>
wrote:</div>
<br class="Apple-interchange-newline">
<div class="">
<div class="">To switch calls from one server to another you
have to use redis and rptengine using HA with pacemaker y
corosync.<br class="">
<br class="">
You must have two OpenSIPs, Two RTPEngine, Two Redis
servers (primary-replica) Two Mariad servers
(primary/primary)<br class="">
<br class="">
With redis you can save calls data (ip, ports, callid) on
active server and then use these data on the replica
server when swithc to active. On my tests, when switching
from a server to another I have between 5 and 10 seconds
without audio.<br class="">
<br class="">
Regards<br class="">
<br class="">
---<br class="">
I'm SoCIaL, MayBe<br class="">
<br class="">
El 19/01/2021 a las 10:00 a. m., Kevin Wormington
escribió:<br class="">
<blockquote type="cite" class="">I’m not using a VIP and I
have made some progress by setting a different active
tag on each node…then upon node failure setting the
failed node's tag to active on remaining node. This
lets the re-invite pinging work, etc. It’s almost there
but the handling of the BYE…they are still sent to the
IP of the failed node even after re-invite pings so any
in-progress calls from the failed node are zombie when
they hang up until the re-invite ping times out (30
seconds). I found an article about initiating a
re-invite on the new node with something like
"opensips-cli -x mi dlg_send_sequential
callid="442CB6C1-6005F8B80009DA08-FC731700"
mode=challenge body=outbound” but that either seems to
terminate the call immediately or say the dialog wasn’t
found.<br class="">
<br class="">
<br class="">
Thanks,<br class="">
<br class="">
Kevin<br class="">
<blockquote type="cite" class="">On Jan 19, 2021, at
8:46 AM, Andy Dierlam <<a
href="mailto:adierlam@ptgi-ics.com" class=""
moz-do-not-send="true">adierlam@ptgi-ics.com</a>>
wrote:<br class="">
<br class="">
With dialog writing to db that both servers use. And
same tag on both - modparam("dialog",
"dlg_sharing_tag", "vip1=active")<br class="">
had this working on opensips 2.4<br class="">
<br class="">
thanks<br class="">
Andy<br class="">
<br class="">
<br class="">
On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <<a
href="mailto:kworm@missouri-telecom.com" class=""
moz-do-not-send="true">kworm@missouri-telecom.com</a>>
wrote:<br class="">
Hi,<br class="">
<br class="">
I've been attempting to get a two node active/active
setup to work with the v3.1 clusterer module sharing
usrloc and dialog. The setup is fronted by a proxy
that handles all of the NAT/media so either OpenSIPS
instance can communicate directly with the user.<br
class="">
<br class="">
What I have working so far:<br class="">
<br class="">
Registrations and calls work when sent to either node
and if you stop OpenSIPS on a node new calls work fine
using the other node.<br class="">
<br class="">
What I can’t get to work:<br class="">
<br class="">
Calls that are already in progress to switch between
nodes when one node fails.<br class="">
<br class="">
<br class="">
I have messed around with various sharing tags…no tag,
same tag, different tags but haven’t had any luck.
I’m guessing that I’m missing something to trigger
the remaining node to send re-invites. Has anyone
attempted this type of setup and have any ideas?<br
class="">
<br class="">
Thanks,<br class="">
<br class="">
Kevin<br class="">
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Users mailing list<br class="">
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class="">
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