Setting up new Asterisk / FreePBX system help

I have also wrote a basic install guide, you could skim that for what ever feature you tried to change that broke things. Might help.

I think one of the things that is causing me so much confusion is that many of the tutorials discuss setting up a FreePBX server locally. I am working with an old FreePBX server running FreePBX 2 and it is a remote machine and our new machine is hosted in a different location but also remote running FreePBX 13.

A lot of the things I am seeing talk about local networking which does not apply to my particular situation. Does this really add a lot of complexity to my situation or am I over thinking the problem and creating my own hurdles that are unnecessary?

Endpoint manager is one of the things I need to get working but dont quite understand how that works when our local network is where most our phones are located with a few located in other countries. The FreePBX server is hosted on a VM in Texas. How does that work?

Thanx again

With endpoint manager, you just have to setup the mac addresses manually, set the tftp or http provision allow in the firewall.

You will have to do something on the phones the first time to point Bremer to your server for the provisioning, but afterwards, they will have the info always short of being defaulted again.

Users will have to power cycle phones to force a reprovision or you will have to remote log in and force the reprovisin if changes were made and are needed instantly. But honestly after initial setup, that is generally rare.

I have a system hosted on Vultr, and I manually created the CFG files for my Yealink phones. I uploaded them to the /tftpboot folder and then in the responsive firewall, I set https provision to other, tftp to reject and http to reject.
I put the remote site IP in the network section as other. I set the. Port assignment in system admin to have https provisioning on default 1443.

I logged into the Yealink phone directly and went to the provisioning tab and told it the URL:

Rebooted the phone and it pulled the config.

The newer model Yealink phones come with a one touch service. I have not played with that yet. But I assume those are like the Sangoma phones. You log into the portal and tell it where to find the provisioning server there.

Basically you always have to have some initial method of telling a new device how to find the provisioning server the first time.

You could do DHCP option, all kinds of things.


Can I get support for this? I am willing to get on the phone with me and help me get this up and running.

Yes, just buy credit and open a ticket:

Your deployment ID is shown in Admin, System Admin, Activation or from the CLI with:

fwconsole sa info

Once you have the Deployment ID, open a customer service ticket with your user details and dep ID, so we can get things straightened out.

I have my deployment ID, the system will not let me create a customer service ticket with my user details and my deployment ID. That is my issue. When I enter my deployment ID for it to search for it says it cannot find it.

I really need to get moving on this and need support ASAP.

Create a new ticket of type “Customer Service and Billing”, you do not need a deployment ID for this.

Done, thank you!

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Given that this is a commercial situation where failure is not an option, you are 100% right to opt for commercial support.

Having said that, it will be worth your while to set up a small server on your own and play with it so you can get a feel for how the system works and build your skill set.

The cloud server portion of this project really adds another layer of complexity, most of it security and NAT related.

My recommendation is to grab an old desktop machine (anything with a p3 or better, ideally newer) and set up a machine in your home. Connect it to an ITSP and a few SIP phones. Make it work, play with the features, and generally geek out. Not only will this help you learn how to manage the system without breaking it, you’ll end up with something in common with the CEO, never a bad thing.

Well with some support from the people at FreePBX I was able to make a lot of progress in getting the system setup and moving toward deployment. I have a lot of setup to do with things like routes, ring groups, conferences, and users before I can deploy but I am very close.

I am having an issue now that I am trying to figure out. It seems that other people have experienced this issue but I have not been able to find a specific way to fix the issue. Outbound calls are dropping at 15 minutes exactly. I see that it is probably related to session-timers=refuse or something with NAT but I have not been able to figure out exactly how to add the session-timers setting to the system so it quits dropping calls.

Any advice here?


Your router is dropping your connection after 15 minutes. There is a setting that you can turn on that will make the router aware of your traffic - I just can’t remember which one it is.

This is a well-known configuration problem that we’ve solved quite a few times. Try looking for ‘15 minute’ in the search bar and see if you can find it.


It doesnt seem like it would be the router. This is happening in multiple locations not just one. It was working fine with our old service running the older version of FreePBX. The new version however is doing this.

I have looked a ton for this issue and have found many answers, a lot of them seem to point to a session timer but I added that yesterday and that doesnt seem to have fixed the issue either.

I will pick through my router just in case


It’s not a router setting It’s an Asterisk setting. The default is something like 3600 seconds, which is too long for the router to maintain the session. Dropping it to 600 seconds solves it. It seems to me it might have something to do with “QUALIFY=” on your trunk.


So I looked at the sip settings in asterisk and dont see anything that would relate. Here are a couple shots of what I see.

What would I need to change in Asterisk and how would I go about doing that? This seems to be the crux of my issue.

Obvious question - did you see this thread:

It speaks directly to the settings that need to be modified to keep this from happening.

Yes, I have read through that thread a few times. I have done the Session-timers=refuse part and that did not make a change. I see some people noted that on asterisk the file /etc/asterisk/pjsip.conf could be modified but I have yet to be able to figure out how to do that through the CLI.

Log into the system (using the console or SSH in from the local network) as root.

vi /etc/asterisk/pjsip.conf

will allow you to change that file. Note that this isn’t a permanent solution - FreePBX will probably overwrite this file on the next “Apply Changes”.

Is there any way you can switch back to Chan-SIP for this? If so, you might find that it works better for these kinds of “external” connections.


This does not seem like a reasonable fix to an obvious issue with this system. There must be a more permanent fix that is not going to require me to either change the technology I am using or to have to rewrite a file constantly.

We picked this system specifically so we could use pjsip, going to Chan-SIP is just not worth it. We might as well stick with our antiquated system that works ok.

How is everyone else fixing this problem?


You are aware that PJ-SIP, while being the SIP technology goal, is not a mature technology yet. Using a hammer when a screwdriver is a better choice is doable, but may not be the best tool for the task. In the specific configuration you have, Chan-SIP may be a better tool for this specific application. Leave the reset of the system as PJ-SIP and add Chan-SIP in for this specific problem space. PJ-SIP is touted to be the replacement for Chan-SIP, but right now, it’s just another way to get where you’re going.

There are lots of other things PJ-SIP doesn’t do well (or at all) so I usually recommend to my customers that we use PJ-SIP for the places it makes sense (like when setting up phones which might need Simultaneous Line Appearance), but for trunks and other places that are set up to deal with the mature technology that Is Chan-SIP, the best choice may well be to use both - after all, there’s nothing preventing you from using one, the other, or both.

Another possibility is that you may need to modify the pjsip-custom.conf file, which will survive a system config rewrite. I think PJ-SIP supports that, so you could try that and see if your changes solve the problem AND survive a rewrite.