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<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 class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS eBootcamp 2021
<a class="moz-txt-link-freetext" href="https://opensips.org/training/OpenSIPS_eBootcamp_2021/">https://opensips.org/training/OpenSIPS_eBootcamp_2021/</a></pre>
<div class="moz-cite-prefix">On 12/30/21 2:50 PM, Sergey Pisanko
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CANbQ6FKuXhExp6-UY1F4jd0GArQba6zUHS3bFgNqacnd5psePQ@mail.gmail.com">
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<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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