New to asterisk and Free PBX and Asterix Now and all this interesting stuff … I was able to create extensions, sip accounts for 2 softphones (x_lite) to ee how this whole thing works on Ubuntu. All is well, with the exception that when dialing from extension 1 to extension 2 (or viceversa), no voice travels in between the two extensions. The phone rings, I can accept the call, but there is no voice.
What I can do (among other things):
a) I can dial in voicemail and check my mail
b) I can dial extension2 extension1 (or viceversa) and leave a voicemail
Any idea of what can be configured wrong in the config files?