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13 to 14 upgrade

October 1, 2018, 3:37 pm
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I am trying to upgrade 13 to 14. I have Sysadmin Pro but the upgrade option is unavailable in there. So I found the upgrade script from the wiki page.

When trying to run Yum I get the following error. Anyone have any suggestions?

 yum -y install http://package1.sangoma.net/distro-upgrade-1807-2.sng7.noarch.rpm
Loaded plugins: security
Setting up Install Process
distro-upgrade-1807-2.sng7.noarch.rpm                    |  12 kB     00:00
Examining /var/tmp/yum-root-olsewb/distro-upgrade-1807-2.sng7.noarch.rpm: distro-upgrade-1807-2.sng7.noarch
Marking /var/tmp/yum-root-olsewb/distro-upgrade-1807-2.sng7.noarch.rpm to be installed
Resolving Dependencies
--> Running transaction check
---> Package distro-upgrade.noarch 0:1807-2.sng7 will be installed
--> Processing Dependency: schmooze-release for package: distro-upgrade-1807-2.sng7.noarch
--> Finished Dependency Resolution
Error: Package: distro-upgrade-1807-2.sng7.noarch (/distro-upgrade-1807-2.sng7.noarch)
           Requires: schmooze-release
 You could try using --skip-broken to work around the problem
 You could try running: rpm -Va --nofiles --nodigest
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Would it be possible to change PABX settings via CLI or changing files manually?

October 1, 2018, 3:37 pm
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FreePBX contains many “custom” configuration files (initially empty), which allow the user to write custom Asterisk dialplan and make other configuration changes, without conflict with the files modified by the GUI. Go to Admin -> Config Edit to see a list of these files.

Also, see Admin -> Bulk Handler, which provides an easy way to add, delete or modify a group of extensions, DIDs, etc. using a script or Excel formula or macro.

If what you’d like to automate is beyond the scope of either of these mechanisms, you could consider:

  1. Look at the PHP code for the module in question and write a script to call the underlying functions.
  2. Learn how the database is modified by the relevant module and perform calls to update the database directly.
  3. Write code to access the web interface, emulating what a browser would do to perform the desired modifications.

Unfortunately, all of the above are at risk of breaking when modules are upgraded.

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Backup can't access certain files due to permissions

October 1, 2018, 3:52 pm
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Your post is basically a red herring :slight_smile:

fop2 runs as root, it’s home folder is /root/, root has supercow powers, you are conflating everything here, /usr/local/fop2 is the location where fop2 stores it’s state there is nothing I said anywhere about users home folder, if you don’t wont to do the change , then don’t and everything will continue to work but the backups, if you DO the changes, then everything will continue to work with the additional fact that the backups WILL start to work also.

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13 to 14 upgrade

October 1, 2018, 4:17 pm
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Phones (Yealink) not registering after PBX reboot

October 1, 2018, 4:18 pm
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In normal operation, when a REGISTER request is sent to the server, the router records an association that directs the replies to the proper LAN address and port. When there is no further traffic for some time, the association is deleted. The phone can also send ‘keep-alive’ packets to prevent deletion; if you want to try a long Registration Retry interval, I forgot to mention that you should also set Keep Alive Type to Disabled. In addition, if your extension is set qualify=yes, the probe packets sent by Asterisk also serve to prevent the connection from timing out. As a result, an incoming INVITE looks like another ‘reply’ to the router and is passed to the phone. (Keep Alive shouldn’t be needed if Server Expires is shorter than the router’s timeout.)

My PBX also has a slow memory leak that I never tracked down. It has swap space that starts to fill once memory is exhausted. Upon seeing that, I reboot it manually, typically about every 3 to 6 months. I suspect the trouble is something not directly related to Asterisk or FreePBX; fwconsole restart does not help. I have Yealink phones in four places; none lose registration as a result of a PBX reboot. Routers are Mikrotik, TP-Link, an ISP-supplied Cisco EPC3925, and an ISP-supplied Neufbox.

For packet capture on the LAN side of the router, the capture feature built into the Yealink phone is probably adequate, or you can capture with Wireshark on a PC connected to phone and router via a managed switch or an old dumb 10 Mbps hub. Or, run the phone traffic through the PC by using two bridged NICs.

You won’t be able to capture on the WAN side unless such a feature is built into the router, because the connection between the WAN interface and the DSL modem is completely internal. However, it’s IMO very unlikely that your ISP is causing the trouble, so running tcpdump on the PBX should give you the same as what you would see on the router WAN side.

If you catch Asterisk sending a reply to a port other than the source port of the request, try setting RPort on the phone for the account in question.

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Delete faxes in /var/spool/asterisk/fax folder older than 30 days

October 1, 2018, 4:20 pm
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Here’s my beta version of the script. I noticed that there were quite a few files left after I ran it that were older than 30 days but didn’t have reference in the db so I’m thinking of adding a find with delete at the end to clean those up:

#!/bin/bash
#
# This script will remove all faxes from the Asterisk DB and the file system that are older than 30 days
# Written by Tim Gross
#
mysql -h localhost -D asterisk -NBe “SELECT faxid,file FROM fax_store WHERE date < UNIX_TIMESTAMP(DATE_SUB(NOW(), INTERVAL 30 DAY));” | while read -r faxid filename
do
tiffilename="/var/spool/asterisk/fax/"basename $filename | sed 's/.pdf//'".tif"
rm -f $filename
mysql -h localhost -D asterisk -NBe “DELETE FROM fax_store WHERE faxid = ‘$faxid’;”
# mysql -h localhost -D asterisk -NBe “SELECT * FROM fax_store WHERE faxid = ‘$faxid’;”;
if [ -f $tiffilename ]; then
echo “<$faxid> <$filename> <$tiffilename>”
rm -f $tiffilename
else
echo “<$faxid> <$filename>”
fi
done

I did the following to clean up the orphaned files and may add these to the end of the script:
find /var/spool/asterisk/fax/ -name “.pdf" -mtime +30 -delete
find /var/spool/asterisk/fax/ -name ".tif” -mtime +30 -delete

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Delete faxes in /var/spool/asterisk/fax folder older than 30 days

October 1, 2018, 4:27 pm
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Probably not a problem in this case, but be very careful with ever using “.” in a sed ‘substitute’ it will match ANY character , you should use "\." to match the literal “.”

Same with “find”

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Would it be possible to change PABX settings via CLI or changing files manually?

October 1, 2018, 5:15 pm
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Hello Stewart1, thank you very much for your response.
Really the interest is in modifying the files so that the changes appear in the FreePBX interface.
Apparently the best alternative would be to call PHP functions.
I studied the basics of PHP without getting into object-driven, but I will try to understand the modules and functions.
Thank you!

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Would it be possible to change PABX settings via CLI or changing files manually?

October 1, 2018, 5:35 pm
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In FreePBX 15 we have a whole new API system that was designed for this exact reason.

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Backup can't access certain files due to permissions

October 1, 2018, 6:18 pm
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Obviously I didn’t make myself clear. Sorry about that. I was the one who had mentioned the user folder in my initial post. I understand what you mean about fop2 and how that will continue to work after changing the permissions. My follow up question was specifically about another aspect of this issue where asterisk couldn’t back up the home folder of the user I had created to manage freepbx from the CLI.

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Backup can't access certain files due to permissions

October 1, 2018, 6:28 pm
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If you examine your methodology, there is not any good reason that the asterisk user should have access to your “home folder” if you need to do that for any strange reason , that then you should add the asterisk user with read permissions to your user group permissions, but be very careful because the asterisk user is also the user than runs apache2/httpd, think about that and what you have in that directory . . .

A better way might be to copy your needed files into /var/lib/asterisk or /home/asterisk, depending on how you installed asterisk/freepbx

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Backup can't access certain files due to permissions

October 1, 2018, 6:48 pm
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I’m really just trying to include this user in the backup along with its ssh keys.

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Backup can't access certain files due to permissions

October 1, 2018, 6:50 pm
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I thought I just gave you two possible solutions, no? .

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Would it be possible to change PABX settings via CLI or changing files manually?

October 1, 2018, 7:02 pm
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Hello Tony Lewis, Thank you very much for your response and I would like to congratulate you and the whole team for FreePBX!
The API news is simply wonderful!
Is it possible to use FreePBX 15 in production?

It turns out that we are starting a new small business in our region, where we will sell SIP numbers in conjunction with URA + Rows + Extensions for small businesses in my region that need to migrate from old telephony to have VoiP advantages.
And as we will begin with a competitive price to try to reach as many customers as possible, the manual configuration of each customer becomes a little time consuming. So I had the idea of ​​creating a shell script that would automate most of the project and then I was thinking of doing a small system in PHP or JS (so I started PHP and JS) to evolve this script and communicate directly with FreePBX, which would be the chosen orchestrator due to its maturity after all these years of evolution.
We have not yet decided whether the best path would be a great server that would concentrate all clients and would work with your HA module in the future, or if the best way would be to automate the creation of Docker containers for each client.
Your advice would also be greatly appreciated by us.

Thank you one more time!

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Sangoma phone only sends first DTMF tone

October 1, 2018, 7:27 pm
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Thanks, I’ll submit a ticket and respond with the results.

Also, my initial post stripped out the XML around the config options for the Sangoma phone; it should have read like this:
<P79 para=“Account1.DtmfPayloadType”>101</P79>
<P20166 para=“Account1.DtmfMode”>0</P20166>
<P1300 para=“Preference.KeypadDTMFTone”>0</P1300>
<P20116 para=“SuppressDTMF”>1</P20116>
<P20117 para=“SuppressDTMFDelay”>1</P20117>

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Would it be possible to change PABX settings via CLI or changing files manually?

October 1, 2018, 7:48 pm
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@viniciusbarreto , Just to let you know before you get too excited, Asterisk makes for a very bad multi-tenant solution. You might want to look elsewhere.

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PAI and CID on 'common' trunks

October 1, 2018, 9:54 pm
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Hi All,
This is my second FreePBX install, but the first where I can’t solve the problem myself.

PBX Firmware: 12.7.5-1807-1.sng7
PBX Service Pack: 1.0.0.0

As far as I’m aware this is the latest/current version, and I have all the updates applied.

My Trunk provider is Spark Voice Connect (New Zealand).
They have given me 1 account, which has a couple of DID’s attached.

They require me to send a PAI as part of the outgoing call authentication process.

The only way I can do this is to have the extension CID set as the Account number, and have the options in Extension/Advanced set as -
TrustRPID ‘Yes’
SendRPID “Send P-Asserted-Indentity header”

Then I need to set the extension Outbound CID to the account number Spark are expecting, so they get the correct PAI.

This creates a few problems though.

The account number (as needed by Spark in the PAI) doesn’t have any calling rights, so it can’t make a call.

And even when they allow calling rights (temporary for testing), the CID is obviously wrong, as it shows the account number, not the CID for the DID.

I’m really stuck here as I’ve been pulling my hair out for the last couple of days, and I’m starting to feel the guys at Spark are loosing patience with me.

How can I send the PAI of the account, and also send the CID of the DID?

I hope someone can help me, and I hope that I can understand what you suggest or request.

Thank you in advance for your help.

Cheers, Sean

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Cant Enable CW on a newly registered Extension

October 1, 2018, 11:06 pm
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Modify time out for a extension

October 2, 2018, 12:41 am
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Hello,

I would like to make that :
When my extension 2080 call the extension 2081 the ring time is set to 10sec.
But when others extensions call the extension 2081 the default ring time is 40sec.

Is it possible to make that ?

Best regards,
Luca

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Modify time out for a extension

October 2, 2018, 12:56 am
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There’s no straight way of doing this.

What you can do is, setup a ring group to ring 2081 with a timeout of 10 seconds, and set failover to Voicemail > 2081.
Then call the ring group, instead if the extension.

Or… You’ll need some custom code.

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