Is there an up-to-date tutorial for FreePBX installation that includes setting up a SIP trunk?

I tried FlowRoute but their tutorials were old as dirt and disorganized as hell. I managed to get FreePBX running on a virtual machine but I wasn’t able to actually complete any calls, and I didn’t have the technical knowledge to figure out where the problem was (they were 100% not helpful when I reached out to them so I gave up temporarily).

Is there a tutorial that goes the entire way? Like, you got a fresh linux install, here is how to install asterisk and freepbx, but then also here is how to set up phones on extensions and then finally add sip trunks so they can call out and receive calls?

Crosstalk Solutions have done many videos on You Tube. They are very good, I learnt a great deal from Chris Sherwood.


He uses endpoint manager, which is $150 right now. I can’t afford that.

What does that have to do with Trunks? The Endpoint Manager is for phones, not Trunks.

You need to provide some actual details of what your issue is. What tech are you using, Chan_SIP or Chan_PJSIP? What are the actual errors you are getting?

I’m not looking for troubleshooting help, but thanks. has great documentation.
Also, as mentioned above, Crosstalk Solutions has great videos, he recently started a v14 101.

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Unless you are using Sangoma or certain Digium endpoints.

I assume that your PBX is running and calling from one extension to another works properly. If not, fix that first.

Flowroute can use either IP authentication or registration. IP authentication is generally more reliable and a little more secure. If the PBX has a dynamic IP address or an unusual network setup, registration may be required.

Assuming IP authentication and pjsip on the default port 5060:

Log into Flowroute and go to Interconnection -> IP Authentication. Note the values of SIP Proxy and Tech Prefix. If the public IP address of your PBX is not listed under My outbound allowed IPs, add it.

Create a pjsip trunk. On the General tab, fill in the Trunk Name e.g. Flowroute and put your main number (starting with 1 if US or Canada) in Outbound Caller ID.

On the Dialed Number Manipulation Rules tab, fill Outbound Dial Prefix with your Tech Prefix followed by *, e.g. 12345678* if your prefix is 12345678.

On the pjsip Settings tab -> General, set Authentication to None and Registration to None. Fill SIP Server with your SIP Proxy and SIP Server Port with 5060.

On the pjsip Settings tab -> Advanced, fill From Domain with your SIP Proxy. Click Submit.

Create an Outbound Route. Fill in Route Name as desired, fill Trunk Sequence for Matched Routes with your Flowroute trunk. On the Dial Patterns tab, fill match pattern with “XX.” (without the quotes). Click submit.

Click Apply Config.

From an extension, dial 1 800 437 7950. You should hear your caller ID read back. (With this simple setup, you must dial 1 before the area code and number.)

Once you get outbound working, adding inbound will be pretty easy.

You Tube: FreePBX 101 v14 Part 11 - Outbound Routes

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