My setup from the other day is now the preferred, Ubuntu 20.04 + Asterisk 18 + FreePBX 15.
All modules are now working… as far as I can tell and in “single player” mode. Now time for “dual player”, working together!
I’m stuck in the loop and don’t know where to start… But the Trunk need to be connected with one incoming and one outgoing Route. Where is this connection made!? In the Trunk or in the Routes, incoming- and outgoing?
We’re using chan_pjsip for the current Trunk, based on what people suggest on this forum. The manual writes about chan_sip and there’s fields for incoming och outgoing phone calls.
I don’t care what flavor to use as long as it’s working!
I’ve now dome some more research with the manager and apparently we’ve been using Telavox + SIP previously in the history of this company. Same number, same subscription but not pjsip.