Using Freepbx 15.0.16.38 GUI, changing the trunk CallerID source caused the trunk to no longer be found as an endpoint

Security issues filed against chan_sip continue to get taken care of. We also continue to review any work done by others against it, but take a deep long look at changes as so far the past ones have caused regressions. We don’t work on new features or bug fixes for it, and with it being marked deprecated a discussion will occur in approximately 4 years for its removal.

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Within the FreePBX GUI it is simpler than chan_sip. I have yet had a need to go outside of the GUI for a standard SIP trunk to work with PJSIP.

That might be due to the fact that unlike Chan_SIP, PJSIP actually cares about having an endpoint, AOR, etc properly configured. Chan_SIP doesn’t care. You can load Chan_SIP configs up with invalid settings and it will not complain or care. Hell, right now in FreePBX there are settings in the Chan_SIP configs that aren’t even valid Chan_SIP settings. PJSIP is also more strict because it follows SIP RFC more closely than Chan_SIP does.

The funny thing is PJSIP actually gives Asterisk the ability to be built out as a PBX doing things that numerous SIP PBX systems have been doing for over a decade well before PJSIP was introduced into Asterisk.

I talked a bit about why PJSIP configuration is the way it is at AstriCon. You can view that presentation at https://www.youtube.com/watch?v=wCOa04g1c7w in case anyone may not have seen it. It’s not focused on FreePBX but it may still be useful to understand some things about PJSIP.

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I listened to this a few months back. It was a good listen.

I see posts here virtually every week complaining about PJSIP not working properly.

As far as the security of Chan_SIP is concerned, there are always known and unknown security risks with using any software, including Asterisk and FreePBX. I have long been an advocate of ensuring that both Iptables and your firewall are strictly configured to prevent access from any unknown source, and from any known source to any port that the source does not require. Doing so ensures that any unknown security risks are protected against.

If you’re opening your PBX to the world and merely hoping that the security holes in Asterisk, FreePBX, Linux (and all of its associated services) are plugged, you’re making a grave mistake.

Most of this thread is unrelated to OP’s post. All are free (for now) to use whichever driver they want, but if you have to ask, PJSIP is the one to standardize on.

Closing thread. @gguldens56 if you ever figure out what actually happened to cause the issue, please share in a new thread.