<div dir="ltr"><div>Have you confirmed a Record-Route header was added to the SIP messages exchanged?</div><div><br></div><div>Anyway, I would try to enable debug log in the SIP client to see what it is deciding to do when you press the hangup button (ideally we should see the BYE and the address to where the request is being sent to, if it is failing to open a WSS connection with opensips etc).</div><div><br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Jun 16, 2023 at 11:22 PM Pat M <<a href="mailto:miagi_son@proton.me">miagi_son@proton.me</a>> wrote:<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 style="font-family:Arial,sans-serif;font-size:14px">so even after adding the record route it still does not hangup and send the bye</div><div style="font-family:Arial,sans-serif;font-size:14px"><br></div>
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------- Original Message -------<br>
On Thursday, June 15th, 2023 at 11:02 PM, mayamatakeshi <<a href="mailto:mayamatakeshi@gmail.com" target="_blank">mayamatakeshi@gmail.com</a>> wrote:<br><br>
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<div dir="ltr"><div dir="ltr"><br></div><br><div class="gmail_quote"><div class="gmail_attr" dir="ltr">On Fri, Jun 16, 2023 at 2:05 AM Pat M <<a href="mailto:miagi_son@proton.me" rel="noreferrer nofollow noopener" target="_blank">miagi_son@proton.me</a>> wrote:<br></div><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><div style="font-family:Arial,sans-serif;font-size:14px">See below</div><div style="font-family:Arial,sans-serif;font-size:14px"><br></div><div style="font-family:Arial,sans-serif;font-size:14px">but what i can see there is no Allow, not sure if that is why?</div><div><blockquote type="cite"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><br></blockquote></div></blockquote></div></blockquote><div>You are not using Record-Route.</div><div>This is not required for SIP to work as long as the UAs are able to talk to each other independently of the SIP proxy after a session is established.</div><div>But probably this is not your case.</div><div>Please try adding this to your route{}:<br><br></div><div> if (is_method("INVITE"))<br> record_route();</div></div></div>
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