Setting up SIP and PJSIP debug

This may seem like a trivial task but for me who knows little of VOIP/SIP its a major undertaking.

I have jitter and dropped calls occasionally and after lots of reading I think I need to set sip set debug on and pjsip set logger on. The challenge is understanding how to do this.

I have FreePBX 15 installed on a little Pi4 and can easily SSH into the unit. I though you access the cli with fwconsole but when you type it on its own, it will just dump the list of commands. I thought that there is a interactive mode? How do I set up the debugging so that I can understand why I’m getting jitter?


From the bash prompt you can get to the Asterisk console (verbosity 3) using:

fwconsole -rvvv

You should also take a look at this forum post that goes into troubleshooting steps you can take on Linux for network-related issues

From a root shell prompt, type
asterisk -r
to get to the Asterisk CLI.
You can then type
pjsip set logger on
sip set debug on
You won’t need both, unless you have a mix of pjsip and chan_sip extensions and trunks.

Some jitter is normal and inevitable. Excessive jitter that causes choppy voice or other quality issues is not and should be tracked down, but pjsip logger and sip debug are not the right tools for that. Also, excessive jitter does not normally result in dropped calls, though if the remote party finds the voice unintelligible and hangs up, then of course the call will drop.

Which calls are affected (calls between extensions, or only external calls)? Does the problem affect what the extension user hears, what the remote party hears, or both? Is it apparently related to activity on the PBX (only occurs when several calls are in progress), on other activity on your network (occurs when watching Netflix in 4K), or something else of which you are aware?

Please describe your setup (devices, internet connection, etc.) and the quality and other issues you are having.

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