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<tt>Hello Sammy,<br>
<br>
Could you check if the BYE you receive does have two RR headers
(one UDP and one TCP) ?<br>
<br>
Regards, </tt><br>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 07.10.2016 15:12, SamyGo wrote:<br>
</div>
<blockquote
cite="mid:CAJUJwthSH57sgxtPoHzgVv8UtpSPsRDz9WuQmZd9tf4ePsE8XA@mail.gmail.com"
type="cite">
<p dir="ltr">Yes record route headers are set just as the default
config file has them. </p>
<p dir="ltr">Also do note that the A party and B party are not
registered users. The setup is also behind NAT as well.
Regardless of these two the calls work completely just flawless
in case the caller side is on UDP. </p>
<p dir="ltr">I do see BYE reach my opensips and processes inside
the In-dialog condition and sent to the correct IP:Port of the
caller just with wrong transport. </p>
<p dir="ltr">I'll post config snippet and traces shortly. </p>
<p dir="ltr">Regards,<br>
Sammy</p>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Oct 7, 2016 2:05 AM, "Johan De
Clercq" <<a moz-do-not-send="true"
href="mailto:johan@democon.be">johan@democon.be</a>>
wrote:<br type="attribution">
<blockquote class="quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div>Bye is sent directly if not record routed. Do you
have record route header?<br>
<br>
<div class="m_-3694584269767684163acompli_signature">Get
<a moz-do-not-send="true" href="https://aka.ms/o0ukef"
target="_blank">Outlook for iOS</a></div>
<br>
</div>
<div class="elided-text"><br>
<br>
<br>
<div class="gmail_quote">On Fri, Oct 7, 2016 at 12:47 AM
+0200, "SamyGo" <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span>
wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="3D"ltr"">
<p dir="ltr">Hi All,</p>
<p dir="ltr">I have a opensips 2.2 with
residential script loaded. A TCP client makes a
call and that call gets forwarded to FreeSWITCH
over UDP. The call establishes just fine and
everything works smooth untill the B party sends
the BYE.<br>
That BYE comes over UDP and hence opensips tries
to send the BYE to the A side over UDP. Hence as
a result A party's phone stays oncall. </p>
<p dir="ltr">I have to manually go to the
loose_route's BYE section and set the
force_send_socket ($fs) to use TCP. </p>
<p dir="ltr">Is there something that tells
opensips to use same transport as the INITIAL
invite? </p>
<p dir="ltr">Regards,<br>
Sammy<br>
</p>
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