If distro: use the version upgrade module. (Find it in the module admin screen)
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Best instructions to follow to upgrade to freePBX 16 from 15?
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Centos EOL - what is upgrade plan?
Debian and Ubuntu. See Official release date of FreePBX 17?. IncrediblePBX, another open source distro, has been running on Debian 11 and Ubuntu 22.04 for several years. See Incredible PBX Wiki | Products
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Correct update FreePBX 13.0.194.10 to FreePBX 16
Is there a reason you cant make space?
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Correct update FreePBX 13.0.194.10 to FreePBX 16
freepbx backup all to /var/spool/asterisk/backup and I cant change this directory or disk…
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Correct update FreePBX 13.0.194.10 to FreePBX 16
You cant clear up space on the system?
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Correct update FreePBX 13.0.194.10 to FreePBX 16
cant.its so small. may be another way how to backup to another location?
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Correct update FreePBX 13.0.194.10 to FreePBX 16
So you’re saying that when you pull out the right side menu you don’t see the options for “Servers”? Because you need to go into Servers and select the type of server you want to backup to. You can choose FTP, SSH, AWS S3, Email, Local…a Local storage is created by default.
You could put the backup on an FTP server…you can put it on AWS S3…
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Correct update FreePBX 13.0.194.10 to FreePBX 16
I do this but error the same:
Saving Backup 5...done!
Initializing Backup 5
Backup Lock acquired!
Running pre-backup hooks...
Adding items...
rsync: writefd_unbuffered failed to write 4 bytes to socket [sender]: Broken pipe (32)
rsync: write failed on "/var/spool/asterisk/tmp/backup-5/var/spool/asterisk/monitor/SIP-6126-0005bf25-2024-01-15-09-29-04.wav": No space left on device (28)
rsync error: error in file IO (code 11) at receiver.c(301) [receiver=3.0.6]
rsync: connection unexpectedly closed (860332 bytes received so far) [sender]
rsync error: error in rsync protocol data stream (code 12) at io.c(600) [sender=3.0.6]
mysqldump: Couldn't execute 'show fields from `cdr`': Incorrect key file for table '/tmp/#sql_6c2_0.MYI'; try to repair it (126)
rsync: writefd_unbuffered failed to write 4 bytes to socket [sender]: Broken pipe (32)
rsync: write failed on "/var/spool/asterisk/tmp/backup-5/var/lib/asterisk/bin/archive_recordings": No space left on device (28)
rsync error: error in file IO (code 11) at receiver.c(301) [receiver=3.0.6]
rsync: connection unexpectedly closed (243 bytes received so far) [sender]
rsync error: error in rsync protocol data stream (code 12) at io.c(600) [sender=3.0.6]
Whoops\Exception\ErrorException: mkdir(): No space left on device in file /var/www/html/admin/modules/backup/functions.inc/class.backup.php on line 211
Stack trace:
1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/backup/functions.inc/class.backup.php:211
2. Whoops\Run->handleError() :0
3. mkdir() /var/www/html/admin/modules/backup/functions.inc/class.backup.php:211
4. FreePBX\modules\Backup\Backup->add_items() /var/www/html/admin/modules/backup/bin/backup.php:54
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Correct update FreePBX 13.0.194.10 to FreePBX 16
Have you tried just backing up the basic configs? Dont backup CDRs, voicemail, recordings, etc.
I also bet there are old log files that can be deleted to make space.
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Can't connect Voicehost SIP
Hey Everyone, I’ve been having some troubles connecting Voicehost SIP Trunk to FreePBX, I have all the usernames, passwords and IP Addresses correct but it just isn’t working,
I run 2 phone numbers through Voicehost and I want them to route to FreePBX where I have everything setup, I used to use 3CX but they IP Blocked me so I just didn’t want to be on their system anymore.
If you need any data to help out just leave a reply
Thanks!
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Change call destination on "Call Reject"
I have a customer that does not have voicemail on all extensions. When they reject a call on an extension that does not have an active voicemail, the call simply hungs up. Is there a way to “divert” the rejected so that the call is sent to another voicemail ?
If I change the “Mailbox” setting in the advanced tab of the extension options to something like 201@device, would the rejected call be transfered to the 201 voicemail ?
Or maybe do I have to change the “Optional Destinations” to point to another extension’s voicemail ? I’m just not sure what scenario would apply (No Answer, Busy, Not reachable)
Thanks
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Correct update FreePBX 13.0.194.10 to FreePBX 16
That sounds like you have space elsewhere, in which case you can create a directory, elsewhere, move the current contents to that directory, replace /var/spool/asterisk/backup with a symbolic link to the directory, and do your backup.
If you can temporarily attach an additional “disk”, you can mount it normally, move the contents, then unmount it and remount it over /var/spool/asterisk/backup.
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Change call destination on "Call Reject"
You are on the right track. Just change the destination for “Busy” under “Optional Destinations” for this extension.
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1 updates could not be installed automatically
After installing FPBX17 on D12 I get a message on console saying.
1 updates could not be installed automatically.
see /var/log/unattended-upgrades/unattended-upgrades.log
after unattended-upgrades runs the first time on the daily timer/cron. On a new install you can run that command manually to see the message immediately
There is nothing in that log file indicating any problem. After doing an internet search for similar problems I determined that this shows up after unattended-upgrades runs which then creates /var/lib/unattended-upgrades/kept-back. Inside that file is line item ffmpeg. Deleting that kept-back file or removing the line item inside it only removes the erroneous message until the next time unattended-upgrades runs. apt install ffmpeg says ffmpeg is already the newest version (5.1.4-8.sng12).
apt purge ffmpeg gets rid of the erroneous message. apt install ffmpeg brings it back. Commenting out the freepbx repo and installing the newer ffmpeg version from official debian repo also gets rid of the error, so I guess it’s because of that newer upstream version. So far I haven’t found any way to have it ignore that higher upstream version and suppress the erroneous error.
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Accountcode value not flowing in extension_custom.conf?
I’m pretty sure I’m missing something obvious here. I’m trying to build a custom dialplan using accountcode. I tried few tests but any gosubif seemed to fail no matter what. I looked at logs and it seems that when I place a call accoundcode is always an empty variable but I don’t understand why.
Here is the current configuration (I stripped everything down to just the accountcode and being able to place a call):
[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(ACCOUNTCODE is ${ACCOUNTCODE})
exten => s,n,Set(default_cid_ca=<ACTUAL_CID>)
exten => s,n,NoOp(default_cid_ca is ${default_cid_ca})
exten => s,n,Set(CALLERID(num)=${default_cid_ca})
exten => s,n,Dial(PJSIP/${OUTNUM}@Twilio,30)
This is the log output (pasting likely relevant lines only):
4250 [2024-05-27 15:02:25] VERBOSE[8209][C-00000050] pbx.c: Executing [s@macro-outbound-callerid:37] ExecIf("PJSIP/98200-0000007a", "0?Set(CALLERID(all)=200)") in new stack
4265 [2024-05-27 15:02:25] VERBOSE[8209][C-00000050] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] NoOp("PJSIP/98200-0000007a", "ACCOUNTCODE is ") in new stack
This is the export of extensions from bulk handler (plz let me know if any other relevant column is needed):
Cross checked Extensions/Advanced:
Is there any other setting I need to enable for accountcode to be used? Is there anything overwriting this variable to nil? Any other pointer please?
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Accountcode value not flowing in extension_custom.conf?
You need to use ${CHANNEL(accountcode)}
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Accountcode value not flowing in extension_custom.conf?
Thanks. I changed to:
[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(accountcode w channel is ${CHANNEL(accountcode)})
exten => s,n,Set(CHANNEL(accountcode)=TEST)
exten => s,n,NoOp(Sanity check w channel accountcode now is ${CHANNEL(accountcode)})
And got again an empty value?
14104 [2024-05-27 20:27:10] VERBOSE[21587][C-0000005d] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] NoOp("PJSIP/98200-00000095", "accountcode w channel is ") in new stack
14105 [2024-05-27 20:27:10] VERBOSE[21587][C-0000005d] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:2] Set("PJSIP/98200-00000095", "CHANNEL(accountcode)=TEST") in new stack
14106 [2024-05-27 20:27:10] VERBOSE[21587][C-0000005d] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:3] NoOp("PJSIP/98200-00000095", "Sanity check w channel accountcode now is TEST") in new stack
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No audio on some PJSIP endpoints
I know I have not provided every detail needed to figure this out but I have provided a lot of information.
Yes, the issue is likely present on certain “networks”. I don’t have the ability to test with multiple addresses with an ISP’s network so I stated an exact problem. I do have the ability to test using my home network and a cellular connection and can reproduce the issue. Using the exact same phones and networks, connecting to two different severs in the same data center that use the same routes and firewalls, gives two different outcomes. One server results in no audio while another server has no issues, Same routes, same phones, same networks. I use the same SIP settings. Trunk calls have no problem so I did not look at direct media but it is not allowed. NAT is enabled and always has been.
I fixed this, but I would like to understand what is causing the issue to happen.
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No audio on some PJSIP endpoints
thank you for ur information
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Accountcode value not flowing in extension_custom.conf?
That log shows it working.
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