Transfering FreePBX to new hardware with new versiuon of FreePBX

Hello - I hope everyone is doing well.

I have an old Dell hardware running FreePBX version The hardware is about 12 years and now very unreliable to be hosting a critical application such as FreePBX. I recently bought 2, Appliances model CIP 745 PBX from ClearlyIP running ClearlyIP 15.0.37 (which I assume is FreePBX version 15.0.37).

Given that we need the phones working all the time including security which is a 24 hour operation, we have to minimize the VoIP system unavailability. The plan was to setup the the new appliances to make sure all works before we turn off the old server, change the IP address of the new server to that of the old server and expect things to work from there.

To execute the plan we need the two servers running on the same LAN. It could have been a matter of backup of the old server and restore but that turned out not workable because of the different versions of the FreePBX. We ended up having to copy the configuration manually.

During testing the new Clearly IP appliance, we find that we are able to call out but when the called party answers the call, the call drops. We are using Zoiper to make test calls from the new ClearlyIP appliance.
The old VOIP continue to work normally.

We assigned a new public IP for the new appliance and port forward TCP and UDP 5060 and UDP port range 10,000 - 20,000 to the internal IP address of the new clearly IP appliance.

The problem appear to be related to our network. When we placed the PBX to an internal network that uses an LTE network, we were able to complete and sustain a call. This indicated it has something to do with our network. Once that was confirmed, the appliance vendor’s engagement was much reduced thats why I am posting the issue here.

We have been troubleshooting now for almost 3 weeks involving Cisco (our Firewall is a Cisco FirePower series 1120). We have another session with them tomorrow to see if we can see whats going on.

My suspicion is that, there could be a conflict between the old and new FreePBX servers, but the problem remain even when we temporariliy disconnect the old FreePbx… There could also be a conflict of the SIP and RTP ports. We are using the same SIP and RTP port ranges although there are two different public IPs point to the two internal VoiP servers.

Has anyone experienced this situation? I will be happy to hear from you. Thank you so much

Confirm that in Asterisk SIP Settings, External Address and Local Networks are correctly set for the ‘new’ public IP. If you change these, after Submit and Apply Config you must restart Asterisk.

In FirePower, confrim that SIP inspection is disabled for the new setup.

If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
make a failing call, paste the Asterisk log for the call at and post the link here.

Many appliance firewalls will apply a ‘helper’ for SIP that assumes you have extensions inside your network, if your ‘server’ is inside that network things can go sideways, best to avoid any ALG’s , SIP helpers or UDP/5060 connections for any number of reasons.

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Thanks Steeart1 - the SIP inspection was always disable on the Firewall
I am not sure what you mean by checking the Asterisks SIP settings. Would that be for the Zoiper client or the server?
I enabled the logger and found the follow a number of log files generated today in the /var/log/asterisks folder - restapps_out_log, queuea_log,freepbx.log,pmslogs. Is there a particular file i should be looking for?

In the FreePBX admin GUI, Settings → Asterisk SIP Settings.

You might click Detect Network Settings and confirm that the correct public IP shows up. (It’s possible that a misconfigured firewall would send unsolicited outbound requests out from a different public IP than the one you are forwarding to the PBX.)

The log you want is /var/log/asterisk/full, which (with pjsip logger on) should contain both the steps executed by Asterisk and the SIP trace.

Assuming that you can call *43 from Zoiper and get a working echo test, the extension is almost certainly set up correctly and you don’t need to change any settings for it.

Hi Stewart

I have the full logs, but the pastebin is not loading for me. The ubuntu pastebin comes up but I am not sure if that works for you. Let me know if ther is another way to share the log file. I can put it in google drive and share it with you or send you the file via file transfer


BTW: the “*43” works


Just a sanity check but you are not using a same trunk registration for both server, right?

I observed similar issue in the past and It was caused by my carelessness; when I copy all the production config to testing server, I forgot to turn off the trunk, inbound call became very unstable… (The call hanged up one or two rings. The session was forced to be terminated since both PBX are trying to register to the trunk provider. That’s totally silly of me, but it took a couple of hours to figure that out…)

You may want to monitor SIP connection flow with sngrep, it’s simpler version of Wireshark I would say, I personally like it if I want to monitor the flow of particular session.

Can’t talk for CIP appliances. But sngrep is shipped with FreePBX distro and AFAIK, PBXact as well. No need to install it.


Hi Stewart1 - I was able to update the Full Asterisk logs. It was a huge file i only took the last log entries from the file. You can find them in


If this is not what you are looking for please let me know.

I was not allowed to post links for some reason. So had to change the format to disguise the link.

You can post links if you enclose them in the preformated text pattern to look like this.

This log shows only OPTIONS (qualify) requests to clearlyip and the responses, plus two REGISTER requests from extension 101 and the replies. There are no calls attempted, in or out.

Hi Stewart1

The full log file is large and when I tried to paste it all, there was a limit to how many characters can be pasted. Is there a specific set of logs I should be looking for? do you have an example?
Is there another way I can send you the file?


Thank you. The new server is using a different trunk setting from the existing PBX. I was using tcpdump but will try the sngrep

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With pjsip logger on: Note the time. Make a failing call. Paste the Asterisk log (/var/log/asterisk/full) starting from the noted time until the end. Unless your system is being inundated with attempted attacks, the paste should be well under 1000 lines, usually much less.

Thanks Stewart1 - I have pasted a new log into past
pastebin com g0S8gfm5

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