the upgrade from the old to the new system will do it’s best to migrate your configuration to the new format.
We expect that all working configuration that work today will also work after the upgrade. Including the mixing of SIP and PJSIP fields.
If you used a PJSIP configuration option, it will be used.
If you used a chan_sip configuration option, it will be translated.
Anything unspecified will implicitly use the pjsip defaults.
This is the same logic as what is currently applied. If there are exceptions there will be an entry in the changelog describing which part of the configuration we failed to migrate.
After the migration chan_sip configurations will NOT be translated to there pjsip equivalent anymore.
nat=force_rport will not work after the migration. you’ll have to use the PJSIP options which are
Any manual configuration files in /etc/asterisk/pjsip.d/ will NOT be modified.