FreePBX Voicemail Issue using SIP URI Issue

And what is the difference between grabbing it from the Asterisk Console output and the logfile? The timestamps? What does that help with? Even with the verbosity set to 10 and the pjsip logger on, you still have to going to the Logfiles part of the GUI and figure out how many LINES of the logfile you want it to display for you. How many lines will that be? Will the OP know how far back in the logfile to go to get all of that particular call if there is so much noise? Will it be 100 lines, 300 lines? How many lines?

When doing it live with the buffer on you can either scroll back to the point where it shows you jumping into the Console or you can grabs the entire buffer history in a copy. From that point anything in that capture has the test call in it. No guessing where the call is or coming back and saying “This isn’t complete, you missed X part of the call. Go back and get the full call.”

I get there can be a lot of “noise” when doing these kinds of captures but part of the problem with the “noise” is that most the people who are asked for this information have no clue what is actual “noise” vs what is actually part of the debug we need to see. So @MichaelCollis01 don’t worry about the “noise”, we (or at least myself) can filter that stuff out and figure out what is the important stuff.

Until we see an actual debug of this in full effect, it’s a guessing game.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.