I am trying to set up freepbx i have used it in the past successfully however there seem to be major issues setting up a Trunk
I have had a provider set up a trunk for me and send the details, attempted to set it up under chan_pjsip however i have been unsuccessful. there are no logs indicating an issue the trunk shows as being online. i have set up an inbound route pointing to the extension that i want it to go to however all i get on the incoming line is busy signal.
no logs show up in the /var/log/asterisk/full to indicate there are any connection attempts beyond this message
[2025-06-19 12:57:48] VERBOSE[1910] res_pjsip/pjsip_options.c: Contact 61XXXXXXXXX/sip:[email protected]:5060 is now Reachable. RTT: 4.291 msec
(provider and account details removed for obvious reasons.
I have even tried blowing away the installation and redeploying it with no success.
I also found that the firewall service was really troublesome in that in the initial configuration it asked me if i wanted to turn it on, then when it got to the part where it whitelists ips and networks it cut all access off to the box before i could set it up properly and i had to go into the cli and manually disable it.
i have tried with version 16 and version 17.
Im not sure what to try now.
The reasons for obscuring the provider are not obvious, given that they are an important piece of information, potentially allowing people to make use of their knowledge of that provider.
The log message indicates good outbound connectivity. If you are not getting inbound calls, you have failed to provide an address, or you have provided the wrong address. Depending on the answer to the first point, that could mean you have failed to register, in which case there should be log entries, you have not correctly specified your public address, or incorrectly included the provider in your local networks, or set the wrong address a the provider end, for one that doesn’t use registration, or your router is blocking unsolicited input from your provider.
the provider is aatroxcommunications.com.au
There are firewall rules on the default gateway to forward port 5060 to the freepbx server..
I called them up and they confirmed the trunk is set up correctly and i also followed the documentation on their site for configuring the trunk. There are no registration attempts on their side showing
Run sngrep. In another ssh window, run fwconsole restart. Do any REGISTER requests for aatrox appear? If so, select one with the arrow keys and press enter to see details. Report what you see.
In the Advanced settings for the trunk, try turning off Permanent Auth Rejection and retest.
If that doesn’t help, wait for attempts to stop and see if the PBX is otherwise still working, for example can you call between extensions or call *43 echo test?
Note the time when trunk activity stops, then look in the Asterisk log (/var/log/asterisk/full) for errors at about that time.
I cannot help you more. I am newbie too. Just shared what I learned about setting up trunks. I found it by pure luck how to check if trunk is connected.
So far I just set trunks between local PBX. I am jet to go that road of making SIP connection to provider.
Registration has absolutely nothing to do with making outbound calls. Registration tells the far side you are registering to where to send calls/requests. That’s it. Some providers may offer an option to not allow outbound if the registration doesn’t exist but that’s not common.
When you make an outbound request such as a call and you’re using username/pass based authenticate you will be challenged for that authentication regardless of the REGISTER status. What is happening here is that the PBX is checking if the trunk being used is showing Available or not.
We need to see an actual debug of what is happening, not just one line from the log, in order to determine what is going on.
we are not really talking about outbound we are talking about inbound here. if i dial the did number from my mobile phone I get nothing but immediate busy signal and no corresponding logs anywhere. its almost like im dialing a DID that isn’t commissioned. Where do i get the debug information your after?
Yes and they just look at this as if its a problem on my side. they also apparently cant see where the call is going only that the trunk is registered and therefor “it should work”
Im reluctant to throw more money at this by opening a new trunk with another provider too.
ok so i bit the bullet and signed up with another provider to rule out the providors trunk as the issue and it behaved a similar way however the the trunk on the providers side was completely set up by me and i realized that i hadn’t set up routing information on the trunk itself (in that providers portal) as soon as i did that it worked immediately.
So its clear at this point the reason this inst working is the provider hasnt set up routing on their end properly. So i have emailed them detailing my findings so will see what they come back with.