I am brand new to FreePBX.
I have installed Asterix and FreePBX and it seems to work.
But what now.
I would like to move a SIP into my FreePBX but I don’t have a clue how to do it!!
Should I “Add SIP Trunk”?
And what should I parse into the fields?
Maybe a link to a easy to understand guide for newbees. Administration, not installation.
First off, SIP is a protocol (that is natively supported by Asterisk), so your sentence “I would like to move a SIP into my FreePBX but I don’t have a clue how to do it!!” makes no sense unless you meant to say “SIP trunk.” I’ll assume you did, and point you to some possibly helpful links. Some of these are distro specific but trunk setup is the same in all, unless you have installed FreePBX 2.8 in which case I don’t know what to tell you - IMHO they “broke” FreePBX with 2.8, in a misguided attempt to make it more newbie-friendly (and if you are running 2.8 and asking this question, that’s evidence that it didn’t work). The principles are the same but you will have to enter trunk dial rules differently (the hard way).
(Note to anyone from the FreePBX team that read this and finds the above paragraph offensive: I assure you it is intentional. You can ban me forever if you like, and I half wish you would, but I will always think that you f**ked up FreePBX starting with version 2.8, and that whoever came up with the idea to make those changes in the trunk setup obviously has no regard for your existing user base, nor any care that it makes almost all existing documentation outdated. I’m holding back here on saying what I really think of the person(s) responsible for this change, although if I thought it would help I would not hold back. It’s only the pain of switching to something completely new that keeps me from wiping our existing setup and going with FreeSWITCH and FusionPBX, or anything that doesn’t have the stink of FreePBX 2.8 on it.)
http://pbxinaflash.net/docs/ especially http://members.optusnet.com.au/bsharif/contributions.htm
http://www.freepbx.org/support/documentation/howtos/hints-on-route-dial-patterns-and-trunk-dial-rules and other documents at http://www.freepbx.org/support/documentation/howtos
The last one might be the place to start if you are using a specific provider that’s in that list.
My version is 18.104.22.168
Thank you for the links. I will read them through. I use a Danish provider for my phone. “Telsome”. I belive I have to ask them how to configure my SIP trunk right?