No voice in between two SIP softphones


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
c) etc.

Any idea of what can be configured wrong in the config files?

Thank you!


The phones are in the same LAN? The RTP ports are opened correctly -if you have firewall-? Can you show us the output CLI when you don’t hear anything?

The answer to the first two questions is yes (actually I predicted this, so I opened up all ports for test puposes, with no effect). When I get home, I will post the CLI output :smile: