Where do I enter SIP user?


My manager sent me a letter containing this information.
“Try to get Asterisk / FreePBX to work with our office number. Try make a call through the server.”

I have no idea where to put this user account information! In the documentation, I see no obvious place…
Please guide me! What am I looking for!? Routes? Trunks?

If this is a new FreePBX system, I recommend that you first configure at least two extensions and confirm that you can make calls between them.

If you are adding this trunk to an existing system, you need to decide what users will dial to call on this trunk, without disturbing ‘normal’ calls that are using other trunks.

There is an important piece of information missing from the trunk information – the domain name or IP address of the provider’s server. Once you have that information, create a new pjsip trunk.
Fill in these fields, leaving everything else at defaults:

Trunk Name: (as desired)
Outbound CallerID: (your phone number, starting with 013)
Username: (your Name, starting with u013)
Secret: (your Password, starting with t59)
SIP Server: (domain name or IP address of provider's server)

Submit, Apply Config. Then go to Reports -> Asterisk Info -> Registries and see whether the trunk is Registered. If not, report the status.

If you get this far, create an Inbound Route with DID Number and CallerID Number left blank (ANY). Set Destination to a working extension. After Submit and Apply Config, try calling in from your mobile and report results.

For calling out, create an Outbound Route with a match pattern for numbers that will use this trunk. If this is a new system, you might start with
and put the new trunk as the only one in the list of Trunk Sequence for Matched Routes. After Submit and Apply Config, try to call your mobile and report results.

Thank you so much for the reply!
This is a new system, since Monday. I manage to get connection following the instructions in this video, excellent production! https://www.youtube.com/watch?v=rtHFdhCm434

But when applying FreePBX on top of Asterisk all these settings were lost… of-course, since FreePBX is suppose to replace manual management. Fine.

Why are you doing things the hard way? If you ha e no idea what you are doing, you don’t start with unsupported, advanced, configurations.

Of course FreePBX can be installed on any (almost) OS that you can install Asterisk on.

But if you have no idea how either Asterisk or FreePBX work the. You should stick with what you see on FreePBX.org, and learn it.

That does not include installing an OS, installing Asterisk, then installing FreePBX.

For a hobbyist, yes. Because you boss told you? No.

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Hit up Youtube and watch Crosstalk Solutions FreePBX 101 guide. Walk through each step with Chris. It got me 98% of the way to a functioning FreePBX system a few years ago.

Jared! With all respect! I’m a learner, my boss is a pro…! :slight_smile: It’s a work and I try to solve it.

I looked at Crosstalk yesterday. Very educative! Thank you!

The server address and the port is missing above, but that part is solved now. After inserting the settings we should “Apply config”, the red box… And PBX was working for about five minutes until the page returned as before.

My manager told me somethings wrong with the privileges and I shall check it out. Where are these and what should it be? What do I look for!?!

" as i said, in the advanced settings you can choose the sip-driver that is active and loaded, i switched from “both” to “SIP only” - and the problem was gone! "

“SIP Channel Driver”


I made this change

~$ cat /var/log/asterisk/freepbx.log

[2021-08-26 07:50:01] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: Adding fwconsole commands - FreePBX OpenSource Project - Documentation [] []

The entire log is full of these messages! What’s the deal!?

I’ll echo Jared’s words again. FreePBX manual install is for experts only. Every official guide on wiki.freepbx.org states so.

I would strongly suggest you install the FreePBX distro and use PJSIP. Disable ChanSIP!

Non Distro - Upgrade to FreePBX 16

Thank you for this link! I think it’s a good advice. I’ll give it a try.

Installing xmpp
Updating tables xmpp_users, xmpp_options…Done
MongoDB is not installed
Error(s) installing xmpp:

Failed to run installation scripts

Updating Hooks…Done

MongoDB!? PBX is using MariaDB and I would expect the installer to now this…! :slight_smile:

Lol, you are installing the XMPP module which apparently uses MongoDB.

To install the official distro with commercial modules, visit:



GUI - How to use FreePBX upgrader tool

Please find below steps to upgrade your FreePBX 15 systems to FreePBX 16 using “PBX upgrader” tool.

  1. Select “FreePBX GUI → Admin → 15 to 16 Upgrade Tool”

Where do I find this selection “15 to 16 Upgrade Tool”?

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