<html><head></head><body><div>Bye is sent directly if not record routed. Do you have record route header?<br><br><div class="acompli_signature">Get <a href="https://aka.ms/o0ukef">Outlook for iOS</a></div><br></div><br><br><br>
<div class="gmail_quote">On Fri, Oct 7, 2016 at 12:47 AM +0200, "SamyGo" <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
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<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>
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