<div dir="ltr"><div dir="ltr">Hi, Bogdan.<div><br></div><div>Yes, you are right. That's full call's scheme.</div><div><br></div><div>Opensips:48470                                 Asterisk (5062)<br>1 leg ------------------INVITE (RR:5061)------------><br><-----------------INVITE--------------------------------- 2 leg</div><div>2 leg --------------OK (RR:5061)--------------------></div><div><--------------------ACK (Route:48470)------------ 2 leg</div><div>< -------------------OK (RR: 48470) ----------------- 1 leg</div><div>1 leg. ACK From UA1 to Asterisk through Opensips (Route:48470) sent, but dropped.</div><div><br></div><div><br></div><div>Best Regards,</div><div>Sergey Pysanko.</div><div><br><br></div><br><div id="mt-signature">
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                    <span style="color:#777">Sender notified by</span> <br>
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    </td><td><span style="color:transparent;font-size:0">01/05/22, 10:45:28 AM</span></td>

                
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    </div></div><br><img width="0" height="0" class="mailtrack-img" alt="" style="display:flex" src="https://mailtrack.io/trace/mail/2a0c5a183b2d62c295dd0093eb1936a40f55b060.png?u=6916461"><div class="gmail_quote"><div dir="ltr" class="gmail_attr">вт, 4 янв. 2022 г. в 20:44, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>>:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
  
    
  
  <div>
    <font face="monospace">Sergey,<br>
      <br>
      I see OpenSIPS sents to Asterisk in INVITE:<br>
      <br>
      Record-Route:
<a><sip:Opensips_IP:5061;transport=tls;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062></a><br>
      <br>
      but in the 200 reply from Asterisk back to OpenSIPS I see:<br>
      <br>
      Record-Route:
<a><sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062></a><br>
      <br>
      Is asterisk the once changing the port there ???<br>
      <br>
      Regards,<br>
    </font>
    <pre cols="72">Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  <a href="https://www.opensips-solutions.com" target="_blank">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 2021 
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    <div>On 1/4/22 3:11 PM, Sergey Pisanko
      wrote:<br>
    </div>
    <blockquote type="cite">
      
      <div dir="ltr">
        <div dir="ltr">Hi, Bogdan.
          <div><br>
          </div>
          <div>Here is my simple scenario description:</div>
          <div><br>
          </div>
          <div>UA1----Opensips----Asterisk ---- Opensips ----UA2</div>
          <div><br>
          </div>
          <div>Transport protocol doesn't change during this chain and
            it's tls, if I understand you right.</div>
          <div><br>
          </div>
          <div>I attached SIP capture of the call. As you can see, there
            is the dynamic tcp port in the RR hrd of last reply to
            client from which Opensips connected to the Asterisk.
            Instead of one, to which UA1 connected to Opensips (5061).
            As a result, there is a media session between UAs, but only
            for 30 sec, during of which the UA1 tried to send ACK to the
            Opensips, but unsuccessfully for quite clear reason. Is
            there the resolution how to realize this scenario without
            rewriting RR?</div>
          <div><br>
          </div>
          <div>Best Regards,</div>
          <div>Sergey Pysanko.</div>
          <div><br>
          </div>
          <div><br>
          </div>
          <div><br>
            <br>
            <br>
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                    <td><span style="color:transparent;font-size:0px">01/04/22,
                        01:46:49 PM</span></td>
                    <td> <br>
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        <br>
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        <div class="gmail_quote">
          <div dir="ltr" class="gmail_attr">вт, 4 янв. 2022 г. в 11:47,
            Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>:<br>
          </div>
          <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
            <div> <font face="monospace">Hi Sergey,<br>
                <br>
                Manually altering the RR hdr is a receipt for disaster
                :). Somehow I suspect you do not do double RR (as the
                protocol changes for the call). This double RR is
                automatically done (by default) when doing
                `record_route()`. Do you get 2 RR hdrs when routing the
                initial INVITE ?<br>
                <br>
                Regards,<br>
              </font>
              <pre cols="72">Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  <a href="https://www.opensips-solutions.com" target="_blank">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 2021 
  <a href="https://opensips.org/training/OpenSIPS_eBootcamp_2021/" target="_blank">https://opensips.org/training/OpenSIPS_eBootcamp_2021/</a></pre>
              <div>On 1/4/22 11:27 AM, Sergey Pisanko wrote:<br>
              </div>
              <blockquote type="cite">
                <div dir="ltr">
                  <div dir="ltr">Hello, Bogdan, .<br>
                    <br>
                    Thank you for your answer. I've solved my issue
                    recently just rewriting Record - Route header with
                    appropriate port within "onreply route block" by
                    subst function.<br>
                    <br>
                    Best Regards,<br>
                    Sergey Pysanko.<br>
                    <br>
                    <br>
                    <br>
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                            <td><span style="color:transparent;font-size:0px">01/04/22,
                                11:27:07 AM</span></td>
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                  <div class="gmail_quote">
                    <div dir="ltr" class="gmail_attr">пн, 3 янв. 2022 г.
                      в 17:59, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>:<br>
                    </div>
                    <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
                      <div> <font face="monospace">Hello Sergey,<br>
                          <br>
                          Could you provide a SIP capture (and calling
                          scenario) to underline the issue you have ?<br>
                          <br>
                          Best regards,<br>
                        </font>
                        <pre cols="72">Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  <a href="https://www.opensips-solutions.com" target="_blank">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 2021 
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                        <div>On 12/30/21 2:50 PM, Sergey Pisanko wrote:<br>
                        </div>
                        <blockquote type="cite">
                          <div dir="ltr">Hello!<br>
                            <br>
                            I try to realize the next scenario with UAs,
                            Opensips-2.4 and Asterisk.<br>
                            UAs are registered onto Asterisk through
                            Opensips and also - on Opensips if the 200
                            OK is came back from Asterisk.<br>
                            Calls between UAs are relayed to Asterisk by
                            Opensips.<br>
                            This scenario works fine with udp. But it
                            needs to do with tls. And here I have the
                            problem. What happens.<br>
                            Unlike udp, tcp cannot listen its port and
                            create clients connection at the same time.
                            Opensips listens tls port for clients
                            connection <br>
                            whereas it creates dynamic tcp port to
                            connect to Asterisk. As a result, I see that
                            port in Record-Route header in 200 OK
                            addressed to caller.<br>
                            Thus, callers ACK comes to that dynamic port
                            instead of Opensips listened port and
                            Opensips dropped it.<br>
                            And question is how to force Opensips to put
                            right port for caller?
                            <div><br>
                            </div>
                            <div>Regards,</div>
                            <div>Serhii Pysanko.<br>
                              <br>
                              <br>
                              <br>
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                                      <td><span style="color:transparent;font-size:0px">12/30/21,
                                          02:49:47 PM</span></td>
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