@mmishra on the FreePBX development team wrote a blog post about some of our recent efforts to try to continue the migration in FreePBX towards PJSIP at:
As I’m sure many of you are aware, chan_sip in recent major releases of Asterisk (17) has been swapped to the deprecated status. This means that bugs and problems are not being fixed anymore in that module and at some point in the distant future it could be removed. In light of this knowledge, it is important for people begin using and evaluating chan_pjsip (if you haven’t already) as this is the way forward from an Asterisk development perspective. Given that, the FreePBX team is making efforts to make migration to chan_pjsip easier as well.
As always, thank you all for your feedback and support! It takes a large number of developers, contributors, people willing to contribute documentation, people wiling to answer questions on the forums, and in other places in order to give a large project like FreePBX the life that it has. I want to personally thank all of you that help with making all the pieces and parts of our vibrant and active open source community come together (as well as your patience with me as I’ve been getting to know many of you better).
I hope you all are doing well at this difficult time.