from fs_cli:
sofia global siptrace on

and you will see the cleartext sip messages...



On Tue, Jun 16, 2020 at 7:13 PM Babak Yakhchali <babak.yakhchali@gmail.com> wrote:
Hi
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:

 https://pastebin.freeswitch.org/view/f9a09df4#L7

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?

any help would be appreciated
thanks
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OpenTelecom.IT
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