Late to the party here, but we want to move to PJSIP for our trunks. We have been unsuccessful thus far. Our current infrastructure is that FreePBX is trunked to our Avaya Session Manager via ChanSIP with the following configuration:
If no luck, at the Asterisk command prompt type pjsip set logger on
make a failing call, paste the Asterisk log for the call at pastebin.freepbx.org and post the link here. If both incoming and outgoing calls fail, paste one of each.
Since you’re not using registration, it’s important that you config the provider portal to send calls to whatever port pjsip is bound to. You obviously also have to config any nat/firewall devices to forward this traffic.
I see my SIP Server port selection in the Connectivity → Trunks section. Within the specific trunk, under the pjsip settings is the SIP Server port field.
Edit: Are you changing the port for a trunk or the extensions?