I’ve just gone from Asterisk 13 to Asterisk 18, dropping SIP and picking up PJSIP along the way.
The process was mostly straightforward (if a bit painful) and at this point the only thing I know isn’t working is this: When I use the Dial command (the way I am set up) in my dial plan I get no audio.
I am going through another asterisk box that has a PRI card (that box has not been upgraded yet, so it still uses Asterisk 13 and SIP - and the exact same config for the past year or so) so basically it goes like this:
IP Phone dials a number, resulting in the Asterisk box calling Dial(PJSIP/5556667777@pbx2) - > PBX2 just calls Dial(DAHDI/5556667777)
The IP phone is not getting any ringing until I answer my test call at the other end. Dead silence until then. But perfectly good call once answered.
I’m pretty sure this is an endpoint thing, but I tried many settings (and module reload res_pjsip) and nothing’s changed.
When that first PBX was Asterisk 13 and SIP things were fine. but I realize 13 SIP to 18 PJSIP is a big step.
Anyone has an idea for me?
Here is the very basic PJSIP config for the PBX I am dialing through