Sip trunk without freepbx. just asterisk conf files ie /etc/asterisk/*.conf

Posted by user20338 on Server Fault See other posts from Server Fault or by user20338
Published on 2010-06-09T21:59:27Z Indexed on 2010/06/09 22:03 UTC
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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?

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