<html><head><style id="outgoing-font-settings">#response_container_BBPPID{font-family: initial; font-size:initial; color: initial;}</style></head><body style="background-color: rgb(255, 255, 255); background-image: initial; line-height: initial;"><div id="response_container_BBPPID" style="outline:none;" dir="auto" contenteditable="false"> <div name="BB10" id="BB10_response_div_BBPPID" dir="auto" style="width:100%;"> Hi Social,</div><div name="BB10" id="BB10_response_div_BBPPID" dir="auto" style="width:100%;"><br></div><div name="BB10" id="BB10_response_div_BBPPID" dir="auto" style="width:100%;">I'll try my luck with Google Translate if you want to share your tutorial.</div><div name="BB10" id="BB10_response_div_BBPPID" dir="auto" style="width:100%;"><br></div><div name="BB10" id="BB10_response_div_BBPPID" dir="auto" style="width:100%;">Thanks, </div>                                                                                                                                      <div name="BB10" id="response_div_spacer_BBPPID" dir="auto" style="width:100%;"> <br style="display:initial"></div> <div id="blackberry_signature_BBPPID" name="BB10" dir="auto">     <div id="_signaturePlaceholder_BBPPID" name="BB10" dir="auto"><p dir="ltr">Kevin V.<br><br></p></div> </div></div><div id="_original_msg_header_BBPPID" dir="auto">                                                                                                                                             <table width="100%" style="border-spacing: 0px; display: table; outline: none;" contenteditable="false"><tbody><tr><td colspan="2" style="padding: initial; font-size: initial; text-align: initial;">                           <div style="border-right: none; border-bottom: none; border-left: none; border-image: initial; border-top: 1pt solid rgb(181, 196, 223); padding: 3pt 0in 0in; font-family: Tahoma, "BB Alpha Sans", "Slate Pro"; font-size: 10pt;">  <div id="from"><b>From:</b> users@lists.opensips.org</div><div id="sent"><b>Sent:</b> January 19, 2021 11:03 a.m.</div><div id="to"><b>To:</b> kworm@missouri-telecom.com; users@lists.opensips.org</div><div id="reply_to"><b>Reply to:</b> social@bohboh.info; users@lists.opensips.org</div><div id="subject"><b>Subject:</b> Re: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure</div></div></td></tr></tbody></table> <br> </div><!--start of _originalContent --><div name="BB10" dir="auto" style="background-image: initial; line-height: initial; outline: none;" contenteditable="false"><div><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">---
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">
      
      <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="color:rgb( 0 , 0 , 0 );letter-spacing:normal;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration:none;word-wrap:break-word" 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="">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="">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="">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="">
                  _______________________________________________<br class="">
                  Users mailing list<br class="">
                  <a href="mailto:Users@lists.opensips.org" class="">Users@lists.opensips.org</a><br class="">
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br class="">
                  _______________________________________________<br class="">
                  Users mailing list<br class="">
                  <a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br class="">
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br class="">
                </blockquote>
                <br class="">
                _______________________________________________<br class="">
                Users mailing list<br class="">
                <a href="mailto:Users@lists.opensips.org" class="">Users@lists.opensips.org</a><br class="">
                <a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br class="">
              </blockquote>
              <br class="">
              _______________________________________________<br class="">
              Users mailing list<br class="">
              <a href="mailto:Users@lists.opensips.org" class="">Users@lists.opensips.org</a><br class="">
              <a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br class="">
            </div>
          </div>
        </blockquote>
      </div>
      <br class="">
    </blockquote></div>
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