Hello, we have a local Freepbx Server that we use to run a phone system at our office. Over the weekend the power went out, when I returned to the office on Monday and turned it back on I found that none of the phone would connect to the system. This was strange to me as in the past restarting the server would usually fix issues with phone not connecting. At this time we are operating off of a failover phone located in the office, but it is preferred if our actual phone system could be working again.
I eventually found on the Asterisk Info screen that all of the extensions and our trunk that are set up as PJSIP are showing offline. I am assuming this has something to do with the issue at hand, but I am unsure. See the picture below for reference.
I have attempted a handful of troubleshooting techniques that I have found online and none of them have worked. I apologize in advance as I am not overly familiar with FreePBX or Asterisk.
Any assistance or troubleshooting tips will be greatly appreciated.
Look in the Asterisk logs for clues about why the registration is failing. Do you see the phone registration attempts? What do the phone endpoint logs say? Any network/Firewall changes that might have caused this? Make sure Fail2Ban is not blocking IPs.
Some of my troubleshooting was specifically pointed at solving the âUnable to retrieve PJSIP transport â0.0.0.0-udpââ error, I had no luck with that.
Sorry but I am unsure how to get to the endpoint logs.
The original problem started before any Firewall Changes would have been made. I tried messing with some of the Firewall settings during my troubleshooting (it has worked for other issues I have had before) but nothing worked there either. Perhaps I didnât try the correct things.
Sorry but I am unsure how to see Fail2Ban. On my dashboard the Fail2Ban has a fire next to it, could this be the issue?
Sorry again for my overall lack of ability with this system, the boss sort of just dropped me onto it.
In the SIP Settings, the port was different so I changed my PJSIP Listening Port to 5062 and that matches my trunkâs port and the ports on each of my phoneâs.
Following a reboot of the Server, this has not solved the issue.
Also, I tried turning off the Firewall just to see if the phones would work, this did not work either. To me this points that it is not a Firewall Issue. I have turned the Firewall back on.
As a point of pedantic correct grammatical style, that should really be âi.e.â (a contraction of the latin âid estâ). (But check your preferred dictionary, thesaurus, style guide)
I tried typing in âman sngrepâ and that is well out of my expertise.
In the meantime I had been in communication from someone from fiverr, in working with him we had Inbound Calls and InterOffice calls working for a while, but eventually the same issue has returned. He indicated that it looked like the issue was coming from our provider (FlowRoute).
The following line was reported as being the issue:
â Contact SSA-Flowroute-Trunk/sip:[email protected]:5062 is now Unreachable
I was going to attempt to reach out to FlowRoute Support soon.
sngrep shows the session, if you donât get a reply to the register, then look to your extension provisioning and router, itâs hard to see in the photo but what are 10.0.0,240 and 10.0.0.236 because 236 doesnât seem to get the call
I just checked a few of the individual extension settings, they seem to look for 5062 on SIP Server 1 and 5060 on SIP Server 2, I am guessing having the 5060 on SIP 2 is a precaution for if the system reverts to looking for 5060.
In the Asterisk SIP Settings PJSIP is listening on 5062.
As of just recently, the extensions are apparently working and we are able to receive inbound calls, though I am unsure if this will actually stay working.
It is only Outbound Calls that we are struggling with at this point. I believe communicating with our provider may help us pinpoint the issue with that.
You and only you are responsible for port assignments, the system will never ârevertâ to what you didnât set it to. You and only you are responsible for making sure all your routers are passing traffic transparently between your extensions and your server, most all calls will have two legs, the A leg between the extension and the server, the server is a back-to-back-user-agent, so will open another session to the chosen trunk, (the B leg), this session must also have both SIP (the call ) and SDP ( the media/audio) routed without hindrance on both the A and the B leg.
sngrep will see both these âlegsâ and both SIP and SDP sessions opened and closedâŚ
, but the quick answer is the interface is fully navigable with arrow keys and the enter button.