Setting up new Asterisk / FreePBX system help


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.

That seems like a reasonable way to approach this. One of the main things we want to use from the PJ-SIP is the ability to have extensions with multiple phones as we have a number of people that use phones in many places.

Aside from that feature, I dont see why we cant use the Chan-SIP for the rest. How would I go about setting the system to use Chan-SIP for our outbound calls?

Thanx again

Add it and configure it to one of the “Alternate SIP ports”. After that, you just send the traffic to that port instead of 5060. In the “Advanced Settings” you can turn on SIP. PJ-SIP, or both. Everything’s already there - you just need to turn it on.

Have the inbound routed to your 5160 (5061, 6050,whatever) port on your IP address and Chan-SIP will to the rest.

After that, once again in “Advanced Settings” you should find the SIP configuration tab - go to the Chan-SIP settings and go to town. You may recall that some folks recommended adding some settings - you can do that at the bottom of the Chan-SIP configuration page using a “keywork=whatever” syntax in the additional settings.

You could also change the phone port from 5060 to 5160 (or 5061, etc.). This has the advantage of obscuring the least secure part of your connections - your inbound phone.

It ends up being just like adding any other channel technology - it just sits there unlike you frob the port.

Unfortunately I dont have the foggiest idea how to do that. I will do some research and see if I can figure out how to do what you just said.

Thank you,