New tool to assist converting from SIP to PJSIP | FreePBX - Let Freedom Ring
The tools is great - but please know that it does NOT move the SRTP or the Call Waiting status. We simply export the extensions when starting, and they are still on ChanSIP, using bulk export, then migrate to PJSIP using the tool, then bulk export the newly migrated PJSIP extensions, and now update based upon the ChanSip bulk export of who si SRTP and who has what call wait setting, then reimport with the correct variables and you are done. Please do note the variable names and options are slightly different for SRTP between ChanSIP and PJSIP.
After you figure out one PBX, the rest come easy.
Also, before we start we go to Advanced | SIP Channel Driver and set it to both. After the migration to PJSIP we set it to ONLY PJSIP. We modify the Server SIP Ports under Asterisk SIP Server Settings to have the PJSIP listen on the exact same ports as the ChanSIP - this way no change at the actual phone extension is required. Do not forget to recreate trunks that are PJSIP and delete the ChanSIP ports. Also do not forget you need to command line and issue an fwconsole stop, and then fwconsole start (I have had instances where the fwconsole stop and start does not work right, a reboot cured this).
After doing this, you now are on only PJSIP - which is the only support standard moving forward.
We have done this to over 60 PBXes with great success.