FOP2 upgrade to 2.31.29 breaking WebRTC on FreePBX v14 and v15

After upgrade of FOP2 with command:
wget -O - http://download.fop2.com/upgrade_fop2.sh | bash

Output while upgrading seems fine:
Reactivation Successful

New configuration file /usr/local/fop2/fop2.cfg.new installed. Original fop2.cfg preserved.
New configuration file /var/www/html/fop2/config.new.php installed. Original config.php preserved.
New configuration file /var/www/html/fop2/admin/config.new.php installed. Original config.php preserved.
Done!
Backing up /etc/asterisk/extensions_override_freepbx.conf to /etc/asterisk/extensions_override_freepbx.conf.bak
Done!

Cleaning and updating /etc/asterisk/extensions_override_freepbx.conf ...
Done!

Creating /etc/asterisk/extensions_override_fop2.conf ...
Done!

Reloading asterisk dialplan...
Done!

Generating FOP2 Manager configuration...

Updating buttons for context GENERAL
Removing button PJSIP/992001
Removing button PJSIP/992002
Removing button PJSIP/992000
Removing button PJSIP/992003
Removing button PJSIP/99001
Removing button PJSIP/992006

WebRTC stops working.
I tried multiple times, its always the same issue.

Sidenote: The update seems to remove the 99xxxx users in FOP2.
One will login directly with the User example 2000 not as usual with 992000 WebRTC user.
I could not find a hint, that this is expected behavior after FOP2 upgrade.

**Error message in tail -f /var/log/asterisk/full
**
[2020-11-19 22:13:30] VERBOSE[20374] res_http_websocket.c: WebSocket connection from '172.16.1.229:51350' closed
[2020-11-19 22:13:41] VERBOSE[20584] res_http_websocket.c: WebSocket connection from '172.16.1.229:51914' for protocol 'sip' accepted using version '13'
[2020-11-19 22:13:41] NOTICE[20584] chan_sip.c: Registration from '"2000" <sip:2000@freepbx.develissimo.com>' failed for '172.16.1.229:51914' - Wrong password

We bought 40$ license today for upgrading purpose but have to stick to old version 2.31.25 of FOP2 due to several issues.

Comments

  • edited November 19

    Info: in old versions when WebRTC still was working log output was like:
    [2020-11-19 22:54:58] VERBOSE[6870] res_http_websocket.c: WebSocket connection from '172.16.1.229:41718' for protocol 'sip' accepted using version '13'
    [2020-11-19 22:54:58] VERBOSE[6870] chan_sip.c: Registered SIP '992000' at 172.16.1.229:41718

    So one can see, that the above mentioned removal of the 99xxxx users during FOP2 upgrade to 2.31.29 seems to be related!
    Please give advice.

  • edited November 20

    Hello boys and girls,

    fixing this was a rougth and timeconsuming task.
    How to upgrade from FOP2 2.31.25 to 2.31.29. We do use FreePBX but this should work for other setups too!

    **How not to upgrade FOP2 2.31.25 to 2.31.29:
    **Do not NOT!!!! use the upgrade command:
    ~~sudo wget -O - http://download.fop2.com/upgrade_fop2.sh | bash
    ~~ <== this will remove your 99XXXX extensions and break WebRTC. At least on FreePBX v14 FOP v2.31.25 on centos.

    **How to upgrade FOP2 2.31.25 to 2.31.29:
    **You have to manually download the tar.gz file like:
    sudo -i
    mkdir Downloads && cd Downloads
    wget http://download2.fop2.com/fop2-2.31.29-centos-x86_64.tgz
    tar xvfz fop2-2.31.29-centos-x86_64.tgz
    cd fop2
    make install
    systemctl restart fop2

    Output:
    Reactivation Successful
    
    New configuration file /usr/local/fop2/fop2.cfg.new installed. Original fop2.cfg preserved.
    New configuration file /var/www/html/fop2/config.new.php installed. Original config.php preserved.
    New configuration file /var/www/html/fop2/admin/config.new.php installed. Original config.php preserved.
    Done!
    [root@freepbx fop2]# systemctl restart fop2
    [root@freepbx fop2]# Connection to vm-freepbx closed by remote host.
    Connection to vm-freepbx closed.
    

    Now your 99XXXX extensions are still alive in FOP2... (with upgrade command they are gone... and FOP2 broken) and WebRTC will still function properly with FOP2 2.31.29 at least on FreePBX v14.

    Now i will try to run the upgrade proccess for FreePBX v14 to v15 without breaking "Call History" in FOP2 but that's another issue...

    Good luck with FOP2,
    Raphael

  • Warning: package update of remaining FreePBX v14 packages now removed the 99xxxx WebRTC Devices again and FOP2 broken like before. Sorry too many test scenarios to respect. Will report as soon as I have more information.

  • edited November 20

    Final procedure...
    if you want to upgrade from v14 to v15 and have same issues with FOP2 than we have, than follow this procedure...
    Do not upgrade FOP2 yet... Hopefully you do have a snapshot/backup to restore.
    First:
    fwconsole ma upgradeall
    fwconsole reload
    Install PBX Upgrade Module manually and Upgrade to v15

    Now you can upgrade FOP2 with the manual tar.gz procedure from above...
    How to upgrade FOP2 2.31.25 to 2.31.29:
    You have to manually download the tar.gz file like:
    sudo -i
    mkdir Downloads && cd Downloads
    wget http://download2.fop2.com/fop2-2.31.29-centos-x86_64.tgz
    tar xvfz fop2-2.31.29-centos-x86_64.tgz
    cd fop2
    make install
    systemctl restart fop2

    reload or reboot.

    FOP2 is now 20.Nov.2020 Version 2.31.29 WebRTC still working fop2users 99xxxxx still available. Fine!
    But "Call History" still broken after this upgrade.
    We try to fix that in another thread.

    Info: If you upgrade FOP2 v 2.31.25 to v 2.31.29 before you do module upgrade an v15 upgrade on FreePBX your WebRTC 99xxxx will be removed from fop2users Table in mysql Database and WebRTC will break! So don't do that!

    Good luck with FOP2
    Raphael

  • edited November 21

    Due to multiple issues (WebRTC 99xxxx extensions gone + Call History unusable,...) after Upgrading FreePBX v14 to v15 and FOP2 2.31.25 to 2.31.29 we take a different approach.
    We stop using WebRTC on FOP2 and switch to SIP softphone called jami which is Open Source and comes with a modern look and feel.
    jami integrates perfectly in our Linux Ubuntu/Debian Desktops.
    WebRTC + Switch-Board-Funktionality always has been an issue. Because as WebRTC user all the cool switchboard FOP2 functionality like call pickup etc... was lost. Keeping FOP2 to was it was designed for. Switchboard functionality only.

    jami way far integrates in our desktop and is the only open source modern SIP softphone we know of. Ekiga etc... all those Open Source softphones are outdated... jami the way to go.

    jami softphone + FOP2 switchboard that's how we clean up our horror upgrade scenario.

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