Any wayto upgrade a freepbx 16 to 17?

Upload the file via WinSCP and do a restore via command line.

So, after my new install and my use of restore, my custom extensions are not seen. From outside asterisk returns a 404 and in the asterisk log it says no matching extension.

So, is this broke in the beta or am I doing something wrong. The extension I want to use is in /etc/asterisk/extensions_custom.conf .


Assuming you are referring to a custom dialplan - does the asterisk see your dialplan when you run dialplan show your-dialplan?

if I type asterisk -rx “dialplan show 656@from-internal-additional-custom” I get the correct 656 extension but trying to call it gives extension not found error as well as calling other custom extensions.

Is 656 pre-empted ?

dialplan show 656@

Perhaps your includes are not correct. What do you see when you run

dialplan show 656@from-internal

Well, in both the 16 and the 17 656 is mentioned in several places, but I cannot call any custom extension in 17, and I have several, so its not just 656. If I just do dialplan show 656@from-internal I get the 656 I am looking for. Very strange.

Can you actually post the output of dialplan show 656@from-internal as well as a call trace?

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OK, some more information. No matter where I try to call, even trying to use a trunk or anything, I get logs like the following:

[2024-04-03 04:00:22] VERBOSE[1654][C-00001876] netsock2.c: Using SIP RTP TOS bits 184
[2024-04-03 04:00:22] VERBOSE[1654][C-00001876] netsock2.c: Using SIP RTP CoS mark 5
[2024-04-03 04:00:22] NOTICE[1654][C-00001876] chan_sip.c: Call from ‘’ ( to extension ‘17034754612’ rejected because extension not found in context ‘fromtrunk’.
[2024-04-03 04:00:22] SECURITY[1670] res_security_log.c: SecurityEvent=“FailedACL”,EventTV=“2024-04-03T04:00:22.664-0400”,Severity=“Error”,Service=“SIP”,EventVersion=“1”,AccountID=“17034754612”,SessionID=“0x7fc0540831b0”,LocalAddress=“IPV4/UDP/”,RemoteAddress=“IPV4/UDP/”,ACLName=“no_extension_match”
I don’t have this error when using freepbx 16 and I did a restore from there to the 17 beta.

Any ideas about my last post? I want to keep this topic open, so I can fix the problem.

Are you using chan_sip?

There have been other posts earlier suggesting that chan_sip doesn’t work with freepbx 17 and you need to switch pjsip

I didn’t know that – last time I looked at this it was very complicated to change, but I am willing to do it. Any documentation as to how to do it? You have to change all extensions, trunks?

I think you have to change all extensions and trunks look at this link:

There have been more discussions here:

Well, I tried to convert the extensions, but that did not work at all – if I look in pjsip_transports.conf I see lines like external_media_address=, I have no idea how to fix.
also, the -a parameter did not work, I got a php error when trying to use it.

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