FreePBX Support - Incoming Call Failures [title edit by mod]

I am going a little crazy. Incoming calls are not getting through on VOIP Innovations because my server wants them to authenticate. I am waiting for FreePBX support who seemed mostly clueless. But it is worse I called them using the emergency PIN 4 hours later no callback. Where is the 1 hour SLA. Once I called them they will look into it and call me back.

On PJSIP settings I have username before IP (which I suspect might be the problem) but if I put IP before username my phones can’t authenticate so I’m not sure what you want me to do here.

Is there somebody who can provide professional grade support. I expected it for $1000 from Sangoma but they don’t seem to be living up to their end of the deal.

Sorry if I sound very frustrated. I am…

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This is a bit confusing. What does changing the VI trunk make your phones no longer register? they aren’t related.

As for the incoming calls, you need to have the VI IPs in the Match field of the trunk.

Disable authentication on inbound for trunk.

Tom,

I am talking about the setting in Advanced SIP Settings–PJSIP.

I have the VI IPs in the match filed. But I’m getting the following (according to VOIP Innovations).

Checking my logs for calls to your DID 7325551234, I found this most recent call:

Date/Time Source Destination SIP Release
2025-03-17 06:54:37.000 +972545551234 +17325551234 SIP Challenge Timeout 504 T

In this case, when we are sending these calls to you, your system is expecting us to send authentication information. The EPG that is routing calls to your IP, Company-Regular, is no tan authentication trunk nor is this feature currently active for your account. Because of this, a 504 SIP Challenge Timeout is being returned to us from your network.

Already long disabled.

That doesn’t mean an authentication issue. It means a response wasn’t received in a certain amount of time. Show these configuration settings you are using. Mask any passwords or sensitive info.

Sure. No passwords anyways. IPs is the most serious thing here… And only publicly available IPs for VOIPInnovations.



And all these IPs are in the firewall? Do you even see these calls hit the PBX system at any level?

Please provide more detailed yet anonymized logs from local PBX firewall, upstream firewall, etc. You may also wish to contact your network carrier to see if they are blocking something.

All of them are in the physical firewall, and the FreePBX Firewall. Honestly I admit being not competent enough to know if they hit the PBX. Tell me where to look and I will gladly do it. But I’m NOT a FreePBX expert.

Please review Support Services - Supplying Useful Information.

SSH into the system and run sngrep that will open up a trace for SIP calls hitting the system at the network interface, i.e. before Asterisk, make a call. You should see the call come into the PBX if it is making it past the firewall and to the system. If the INVITE is hitting the system then something else is going on, if you don’t see it then there’s a networking/firewall issue.

Hi @nsumner Sorry you’re having an issue, we’re here to help but I wanted to provide clarification on our POMPs plans for FreePBX.

Our Peace of Mind Packages provide a guaranteed response time based on the classification and severity of an issue.

Gold offers a 4 hours response time during our normal business hours for issues deemed critical, while our enhanced Platinum POMP ($900) offers a two hour response time 24X7

Full details/terms can be found here.

Please let me know if you have any other questions. I’ve dropped you a PM with my direct contact info. Thanks for using FreePBX!

Michael. I added a platinum POMP. It seemed to have little effect. Actually sadly disappointing.

I can see everything (INVITE) hitting the system and being REJECTED. So the there is that. It is indeed hitting the system and being rejected. Very easy to see thank you for that knowledge.

OK then you need to see why it is being rejected. The next step is to SSH back into the system and do the following

asterisk -rvvvvvvvv << this will log you into the Asterisk console and show call processing output
pjsip set logger on << this will enable the SIP debug to show the sip messages.

Do that, make a call then copy and paste that data at pastebin or some other service and give us that link.

Assuming this guide is current, go through and check all those settings.