<div dir="ltr"><div>from fs_cli:</div><div>sofia global siptrace on</div><div><br></div><div>and you will see the cleartext sip messages...<br><br><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, Jun 16, 2020 at 7:13 PM Babak Yakhchali <<a href="mailto:babak.yakhchali@gmail.com">babak.yakhchali@gmail.com</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 dir="ltr">Hi<div>I'm using webrtc with chrome and sipjs as client on freeswitch 1.10. Calling to some endpoints results in NO_USER_RESPONSE. I've not access to client systems and it makes debugging the problem harder. this is the freeswitch log for one of those calls:</div><div><br></div><div> <a href="https://pastebin.freeswitch.org/view/f9a09df4#L7" target="_blank">https://pastebin.freeswitch.org/view/f9a09df4#L7</a></div><div><br></div><div>Does NO_USER_RESPONSE mean that invite is not reached to the client browser or it is reached but no reply is received? can it be related to network problems or NAT?</div><div><br></div><div>any help would be appreciated</div><div>thanks</div></div>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div><br clear="all"><br>-- <br><div dir="ltr" class="gmail_signature">Sincerely,<br><br>Giovanni Maruzzelli<br>OpenTelecom.IT<br>cell: +39 347 266 56 18<br><br></div>