Exciting News: FreePBX 17 is Now Generally Available!

Hi Everyone,

We are excited to announce that FreePBX 17 is now officially available and runs on Debian!

This release marks a significant milestone for us, and we look forward to hearing your feedback. If you encounter any issues or have suggestions, please share them by raising an issue on our GitHub issue tracker.

For more details about FreePBX 17, visit our blog announcement page.

Thank you for your continued support!

Best Regards,
Kapil

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Thanks. With lots of 16 systems out there, I guess its as good a time as any to take the 17 dive.

Just installed Debian 12 on my ESXI server, set static IP and FreePBX install script is running now. See how it goes.

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Kapil,

On Exciting News: FreePBX 17 is Now Generally Available! | FreePBX - Let Freedom Ring you make the blanket statement:

“Note that FreePBX 17 supports chan_pjsip only.”

However on that page you have a link:

  1. FAQs on Freepbx 17

And on that FAQ page the 5th question down is:

" How to use chan_sip with the FreePBX 17.0 ?"

There is also a link on that page to

  1. Official FreePBX 17 site

And the * Frequently Asked Questions

link on THAT page goes to a much smaller FAQ - which is newer - and because it’s smaller - it’s much less useful.

I think it would be much better for you to state on your page:

"“Note that the FreePBX project supports chan_pjsip only on FreePBX 17. However there are two unsupported methods of adding chan_sip into FreePBX 17 + Asterisk 21”

for business users that have PBXAct on official hardware whats the security of having centos 7 aka pbxact 16 on a live connected to the internet service now that centos has been EOL a month ago.

Also when would 17 be production ready?

With correct use of firewalls, none, we have hundreds of clients still happily chirping away without issue on FreePBX 14, including one facility with thousands of users.
Even with HA its not the sort of place that can have downtime to upgrade (unless some construction crew manages to wipe out its diverse fibre links AND take out the microwave mast - heh, maybe then they’ll call us with the go ahead)

i guess the next question is, whats the correct use of firewall. i use the one provided with the pbx set to automatic.

Hello Kapil Gupta,

@kgupta

Please post/update the FreePBX 17 release announcement on https://distrowatch.com.

Clifford

Added one note in the main Freepbx 17 installation page.
https://sangomakb.atlassian.net/wiki/spaces/FP/pages/222101505/FreePBX+17#A-note-for-Chan-SIP-Support-in-FreePBX-17

And thanks for highlighting the FAQ url in our official website , will update that to redirect our main Freepbx 17 FAQ url.

@cgonsalves25 Can you explain why you believe FreePBX should post on DistroWatch? The project doesn’t have its own distro anymore, and is more like a regular application now.

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There’s no immediate need to upgrade if the PBX is not directly accessible from the Internet. However, in theory a virus could be downloaded by a user with a web browser on the “inside” that could find a vulnerability in an older Linux distro such as on the PBX.

If you are not using chan_sip you can easily do a backup and restore from the FPBX 14 to a FPBX 17 system on a test network, make sure phones register, etc. and swap it in.

Kapil,

On the page:
Step By Step Debian 12 Installation - FreePBX Open Source - Sangoma Documentation (atlassian.net)

step 28 you have the command:
apt-get-y install net-tools htop screen tshark vim sngrep

That should have a space in front of the -y, thus:

apt-get -y install net-tools htop screen tshark vim sngrep

Done. thanks @tmittelstaedt

Kapil,

What is your feeling on people making qcow2 VM image files & XML config files available of FPBX 17 with the USECALLMANAGER patch applied to chan_sip, Asterisk 20, and chan_sccp-B applied to Asterisk 21? Obviously not “configured” ones. They would have to change passwords/hostnames/etc/hosts, apply static IP, apply passwords and run qemu-img resize (if they wanted) as well as modify the MAC address (and add/delete user IDs in the /etc/password of the VM)

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there is a few things that didn’t work for me, so I ended up following this guys, much more indepth instructions at https://bit.ly/4dSlbhm

Please don’t use URL shortners since bit.ly could go out of business in the future and then the URL you posted goes nowhere. The actual URL is:

His isn’t a bad effort but it does have an incredibly uninformed rant in it:

"FreePBX 17 installs Asterisk 21, which no longer supports chan_sip or the Macro dialplan application, this could lead to issues if your trunk provider and peers don't support pjsip - they should, the removal of chan_sip has been coming for a long time, it was marked deprecated and unsupported since Asterisk 17 back in 2019, but many years before that everyone was warned it was being retired, it is a very old and limiting protocol, it was further warned Asterisk 21 onwards will have no chan_sip code in it, so if your provider does not offer pjsip, they've been more than just asleep at the wheel and don't deserve anyone's business, Asterisk, Broadsoft, and others, have supported both protocols for a very, very long time, pjsip although first written in 2002, was released in 2005, that's nineteen years ago."

Where do I even start? Sigh.

1) FreePBX 17 offers the ability to downgrade to Asterisk 20 after installation which includes chan_sip

2) chan_sip and chan_pjsip are NOT two different protocols! They both implement SIP

3) "provider doesn't offer pjsip" how the heck would you even know what they are using for SIP?

4) "if your trunk provider and peers don't support pjsip" What does this statement even mean??? The only trunk providers not supporting SIP are the ones delivering the trunks on a POTS line. And even some of THEM will happily switch you over to an Ethernet drop with a sip trunk on it. It's SIP not pjsip. pjsip is NOT a protocol. SIP is the protocol.

5) "it was marked deprecated and unsupported since Asterisk 17 back in 2019" FreePBX is unsupported. Asterisk is unsupported. NONE of this stuff is "supported" If you want a "supported" Asterisk installation you buy PBXact or something from Grandstream or someone else.

6) chan_sip is still available and still being maintained you merely have to issue an extra command during Asterisk compilation to have it included.

7) chan_sccp is also still available and still being maintained and once more you also merely have to issue an extra command during Asterisk compilation to have it included

Clearly he does not know what a channel driver is.

BTW it’s also bad form to make statements like the backup/restore process doesn’t work without at least trying to report the problem back to the FreePBX project.

Good to see your same attitude towards the world on mailing lists carries on here too.
First, bit dot ly has been around a long time, it’s not going anywhere, and if it does, so what?

2, the rant is his opinion saying what he experienced, I’m so sorry it bursts some bubble.
I should suggest he come here to answer your very obvious nose out of joint response, but I’m not so petty that I crave to start a trolling match between you and him, did you comment over there? No? I don’t see it.

It kind of stands to reason that if its deprecated and now removed as stated on asterisk.org then its obviously dead, even I can see that.

As for your nit picking on protocols, I understood what he was getting at, after all I think his article is meant more for clueless, since if you are experienced, you wouldn’t need to read it in the first place - K.I.S.S. look it up.

I cant talk for him and wont, but the BTW at the end, he did say nothing was available for him to see, presumably that means nothing in the log either, since he said where what and where the log was you would imagine he would have looked, so what would you report it as Ted if you have generic banner errors but no substance, we know where it would end up - file 13.

pssssst ted
sshh dont tell FreePBX , seems they call it a protocol too, sshhhh

I never seen any compatibility issue with a SIP provider and PJSIP.
Are you sure a SIP provider uses SIP or PJSIP?
There is several SIP servers based on other drivers outside PJSIP. It should work.