Asterisk 18.3.0 , pjsip

Hi all
I try to communicate througt a sipelia pbx.
i obtain this sip flux
Can you try to explain me this flux?

we need to also see the full trace in order to make sense
as there could be many valid reasons for remote end to send an invite
fx one common reason is the remote want to update name/number, codec selection or session-timers

also you may want to consider updating to the latest minor release 18.12.1
to avoid using time troubleshooting as issues that already has been resolved

Thanks TheMark
i try to get the full trace

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