Sip trunk without freepbx. just asterisk conf files ie /etc/asterisk/*.conf
- by user20338
every damn tutorial uses buggy freepbx. It's a pain in the ass because it doesn't set itself up right. I had to manually intervene and so did others.
I want to be able to take pstn calls using my Linksys SPA3000 as a gateway.
I'm told I need to add a SIP trunk then add it to a outbound route.
Also how the heck is this outbound route within twinkle my sip client?