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Just comment everything in the file /etc/dahdi/modules and reboot. It shouldn't load any DAHDI modules.

Is directed call pickup enabled in the feature codes (Admin -> Feature Codes -> Directed Call Pickup)?

If you can get to the mail logs on the beaglebone, they should tell you right away what the problem is. By default, they should be in /var/log/maillog on a regular CentOS install of PIAF. Not sure where they'd be on the beaglebone.

Another thing to consider if your web server is remote from your PBX, is to setup master-slave replication of the cdr MySQL table to a MySQL server running on your remote web server. Then you only need to allow replication between the two MySQL servers through your firewall and you will have...

Now you need the DateTime Perl package as well, so do a yum install perl-DateTime.

Try doing a yum install perl-MIME-Lite from the linux prompt and see if there's a package for it. I know that with CentOS v6.4, there is a package, but don't know about v5.5.

This looks like an issue with one of the FreePBX modules which was recently updated. See here for more info.

There's a type-o for one of your times, which is listed as 12.30 and should be 12:30. With that change, the SQL should work. Make sure there is no space when specifying the MySQL password (i.e. you should specify your password like so: -pmypassword and not -p mypassword). Also note, that the...

Looks like you need some interactivity to enter the remote password. Try doing this: su - asterisk ssh-copy-id [email protected] and enter the password for [email protected] when prompted. ssh-copy-id simply copies the public portion of the key pair you created on your PIAF box to...

There was a type-o for the directory in my post above, which I edited within a few minutes of posting. The directory should be /var/lib/asterisk/.ssh/ instead of /var/lib/asterisk/

As mentioned above, you need to have already verified the remote system's SSH key. Since you're getting a "Host key verification failed" error, I'm guessing this was not done. You can append the remote host's SSH key to the asterisk user's known_hosts file by running this command as root from...

Asterisk, at least on PIAF, runs as user asterisk. The .ssh directory for the user asterisk is /var/lib/asterisk/.ssh To setup an ssh key for the asterisk user, do this as root from the Linux command prompt: su -c "ssh-keygen -t rsa" asterisk Select the default for the next three questions...

Try this (which may or may not work)... Remove voicemail from the extension that's causing problems through FreePBX. From the Linux prompt, entirely delete the voicemail folder for the extension experiencing problems. For example, if the problem extension is extension 500, and assuming the...

Just forward 5060 on your router and post the SIP credentials for your 50 extensions. It'll be tested for you in no time! :D

wardmundy - any chance that wav->mp3 script can be rolled into the next release of PIAF? It's so useful, it should be part of PIAF, not an add-on.

If you're trying to prevent fail2ban from banning your IPs, you'll need to edit /etc/fail2ban/jail.conf and append your network/IPs to the ignoreip parameter. Remember to restart fail2ban after you make the changes (from the linux prompt: service fail2ban restart).

What version of FreePBX are you running? In FreePBX v2.11, if you go to Settings -> Asterisk SIP Settings, you can set the RTP port range under MEDIA & RTP SETTINGS -> RTP Port Ranges. If you're running an earlier version of FreePBX, you'll need to edit the file /etc/asterisk/rtp.conf and...

Not to discourage using the Paging Pro module, which I'm sure will do the job quite easily through a nice GUI, but you can quite easily accomplish this with a cron and a call file. Construct a call file like this: Channel: Local/9999@from-internal CallerID: <1111> MaxRetries: 1 RetryTime: 30...

You could run this on cron: 02 3 * * * asterisk /usr/bin/find /var/spool/asterisk/voicemail -type f -mtime +45 -delete That will run every day at 3:02 AM and delete any file in /var/spool/asterisk/voicemail that was modified 45 days or more ago.

Yeah, I'm not terribly worried about it since it is working as expected. The calls are not automatically forwarded based on time. They are manually forwarded using a call flow, directed to the Call Recordings module which then forwards the calls to the answering service. This setup has been...