Notes on FreePBX, IncrediblePBX, and the like.

Installation

Copy the ISO to the SDHC card

• Download IncrediblePBX for Pi. See http://nerdvittles.com/?p=3026
• Unpack the tarball.
• Put the SDHC card in the card reader (assumes Linux).
• Run the make-sdhc script that came out of the tarball.

Put the Pi on the network

For security reasons, do not put the Pi outside your firewall!

Boot the Pi.

Put the SDHC card in the Pi, attach the HDMI port to a monitor, and boot.

Initial configuration

• Log in as root. Initial password is raspberry.
• Allow IncrediblePBX to update itself.
• At the bash prompt, run raspi-config.
• If the SD card has more than 4Gb, choose the menu item to expand the file
  system. (The ISO is only 4Gb, and Linux will only see that much, unless
  you expand the file system.)
• Possibly select the overclock option, as well.
• Enable the SSH daemon.
• Reboot.

In a terminal window on another machine, ssh into the PBX machine.

Configure MySQL

The docs I’ve seen don’t say what the IncrediblePBX distribution uses for
the MySQL root password, and it’s useful to be able to get into MySQL. Set
the root password as follows:

# /etc/init.d/mysql stop
# mysqld_safe --skip-grant-tables &
# mysql -u root
mysql> use mysql;
mysql> update user set password=PASSWORD("NEW-PASSWORD") where user='root';
mysql> flush privileges;
mysql> quit
# /etc/init.d/mysql restart
# mysql -u root -p

Change the host name

Edit /etc/hostname to reflect your desired host name. Then, run:

# hostname $(cat /etc/hostname)

Install and configure postfix

This is only necessary if you prefer postfix to the default exim.

# apt-get install postfix

A smart host relay is sufficient. Having a configured mail server allows
the PBX to email voice mail audio files.

Using Gmail as a relay

First, you’ll need TLS support, which requires installing some supplementary
packages.

# apt-get install libsasl2-modules

Next, add the following to /etc/postfix/main.cf

# Use Gmail as smart relay. See
# http://mhawthorne.net/posts/postfix-configuring-gmail-as-relay.html and
# http://anothersysadmin.wordpress.com/2009/02/06/postfix-as-relay-to-a-smtp-requiring-authentication/
relayhost = [smtp.gmail.com]:587
smtp_sasl_auth_enable = yes
smtp_sasl_password_maps = hash:/etc/postfix/sasl_password
smtp_sasl_security_options = noanonymous

smtp_use_tls = yes
smtp_tls_security_level = secure
smtp_tls_mandatory_protocols = TLSv1
smtp_tls_mandatory_ciphers = high
smtp_tls_secure_cert_match = nexthop
smtp_tls_CAfile = /etc/ssl/certs/ca-certificates.crt

Change masquerade_domains appropriately, as well.

Finally, you have to add authentication parameters. Create
/etc/postfix/sasl_password with the following contents:

[smtp.gmail.com]:587 gmailuser:gmailpassword

Replace gmailuser with your Gmail login. If you use a Google Apps
domain account, use your domain email address as the user.

Then, run these commands:

# cd /etc/postfix
# postmap sasl_password
# chown postfix sasl_password*

Adjust your router’s firewall

With your Pi behind your firewall, you’ll have to open some ports. Configure
your firewall to pass UDP ports 5060, 5061, and 10,000 through 20,000 to and
from the Pi. If you’re using a Linux-based iptables firewall, the following
rules should do the trick

EXT_IF=...   # The external, Internet visible interface (e.g., eth0)
PBX=...      # The internal IP address of your Raspberry Pi PBX machine

iptables -t nat -A PREROUTING -i $EXT_IF -p udp -m udp \
         -dport 10000:20000 -j DNAT --to-destination $PBX
iptables -t nat -A PREROUTING -i $EXT_IF -p udp -m udp \
         -dport 5060:5061 -j DNAT --to-destination $PBX

Configure the PBX

• Connect to the web server on the Pi.
• Select the administration logo.
• Log in as admin, with password admin.
• Change the password, under the Admin > Administrators menu.
• Under the Settings menu, select Asterisk Settings.
• Set the NAT configuration.

Configuring Trunks

Google Voice

Log into your Google Voice account. In Settings, set:

• Call Screening – off
• Caller ID (incoming) – Display caller’s number
• Do Not Disturb – off
• Missed Calls – both options should be unchecked
• Call Options (Enable Recording) – off
• Global Spam Filtering – on

NOTE Disable 2-factor authentication on the account, or the PBX won’t be
able to log in.

Next, set up a trunk in the PBX. Under Connectivity, select
Google Voice (Motif). Create a new Google Voice trunk:

Leave all other settings alone.

Note 1: If you’re using a regular Gmail account with Google voice, your
username should be only your username, with no “@gmail.com”. If you’re using
a hosted Google Apps account, specify the username and the domain name.

Note 2: Specify the number without spaces or hyphens.

You’ll need to set an appropriate inbound route. (See below.) You can also
use Google Voice as your outbound route.

Vitelity

Sample trunk configurations, using a subaccount

Outbound

Set Maximum Channels to 2.

Set the Outbound Caller ID to the number for the DID you’re using
for outbound calls.

Set the trunk name under Outgoing Settings (e.g., vitel-out).

Set PEER Details under Outgoing Settings to something like the
following. Replace the username and secret to correspond to the Vitelity
subaccount.

type=friend
dtmfmode=auto
username=subaccount_user
secret=subaccount_password
fromuser=subaccount_user
trustrpid=yes
sendrpid=yes
context=from-trunk
canreinvite=no
nat=yes
host=outbound.vitelity.net

Inbound

Set the trunk name under Incoming Settings (e.g., vitel-in).

Set USER Details under Incoming Settings to something like the
following. Replace the username and secret with the credentials of the
Vitelity subaccount you’re using.

type=friend
dtmfmode=auto
username=subaccount_user
secret=subaccount_password
context=from-trunk
insecure=port,invite
canreinvite=no
nat=yes
host=inbound29.vitelity.net

Set the Register String. Again, replace the username and secret with
the credentials of the Vitelity subaccount you’re using.

subaccount_user:subaccount_password@inbound29.vitelity.net:5060

If inbound calls aren’t working, be sure to set the provider route for the
DID(s) appropriately, via the DIDs page on the Vitelity customer portal.

Configuring routes

Configure your trunks. (See above)

Inbound route

Define a new incoming route and set the DID Number field to the number
associated with the trunk. You must ensure that the trunk passes the associated
DID number, or routing won’t work.

Be sure to define a destination (e.g., a ring group or extension.)

Google Voice Note: If you want to route from a Google Voice trunk, just
create a new route and put the Google Voice number in the DID Number
field.

Outbound route

Define the route as normal, but be sure to use dial patterns so that outbound
calls match the route.

Configuration for a single default outbound route

(Dots in the above indicate blanks.)

Forwarding to an external number after so many rings

Two simple solutions:

Follow Me

1. Applications > Follow Me
2. Select the extension.
3. Select hunt as the Ring Strategy.
4. Add the number, with a trailing #, to the Follow-Me List.
5. Optional: Adjust the ring time.

Note: If the target of an inbound route is a ring group, rather than an
extension, you must include a “#” on the end of the extension in the ring
group to force Follow Me behavior.

Chained ring groups

1. Put the external number in its own ring group (e.g., ring group #601).
2. Set the “Destination if no answer” to “Voicemail”.
3. Put the main extension in different ring group (e.g., ring group #600),
   and point the inbound route at this ring group.
4. In this primary ring group (#600), set the ring time appropriately.
5. Set the “Destination if no answer” to go to the second ring group.

Ensuring that the original caller ID follows a forwarded call

Go into Advanced Settings and change SIP trustrpid and SIP sendrpid to
yes.

Also, make sure you don’t have the Outbound CID set on the primary
extension.

NOTE: This strategy may not work. The DID provider may not honor the PBX’s
attempt to pass along the original caller ID. (Mine doesn’t; I always see the
caller ID of the PBX’s outbound trunk.)

In that case, one solution is to modify the second ring group (#601, in the
example above) so that “Confirm Calls” is checked. When the call is
forwarded, the PBX (which knows the original caller ID) will play a short
message (“press 1 to accept the call, 2 to reject it, 3 to listen to the
caller ID information”). This approach provides a way to allow the callee
to get the caller ID information, even though it’s not passed along by the
telco. Be sure to configure enough ring time on the second ring group to
permit the callee to listen to the message!

Ensuring that an extension only uses specific outbound routes

This is useful if you have, say, a home line and an office line on the
same PBX, and you want to ensure that each one uses a specific outbound route.

Solutions proposed on the web include:

1. Using the complicated, and unsupported, Custom Contexts module.
   This is a potentially dangerous module that can totally bork your PBX.
2. Using something called Extension Routing, by SchmoozeCom; getting it
   requires a complicated registration procedure at SchmoozeCom, one that
   doesn’t appear to work with Incredible PBX.
3. Using a dial pattern to specify individual internal extension caller IDs
   on each outbound route.

(I have not gotten this working yet.)

Modifying the inbound caller ID.

• Install the Set Caller ID module. This module works as a filter.
• Create a new Set Caller ID, editing the inbound caller ID as appropriate.
• Set the new Set Caller ID’s destination to be the desired extension or
  ring group. If adding a prefix (e.g., “H” to indicate that the call came
  into the home line), it’s best just to modify the caller ID name.
• Go to the appropriate inbound route and set its destination to the new
  Set Caller ID.

Hold Music

Converting the music file

FreePBX works best with 8- or 16-bit monoaural, 8,000-Hz WAV files. If
you’re starting with an MP3, first convert it to a WAV file:

$ lame --decode mymusic.mp3 mymusic.wav

Then, use sox to convert it to the correct format:

$ sox mymusic.wav -r 8000 -s -b 8 -c 1 mymusic2.wav

Installing the music file

FreePBX support hold music categories. While the Settings > Music On Hold
screen supports uploading files and creating new categories, it doesn’t always
work. It’s easier to do the whole thing from the command line.

• Copy your converted WAV file(s) to the PBX machine.
• SSH into the PBX.
• cd to /var/lib/asterisk/moh
• If you want the new music to be in the Default category, copy the WAV
  files to that directory. Otherwise, create a subdirectory for the new
  category (e.g., mkdir Classical for a “Classical” category), and copy
  the WAV files into that directory.
• Restart the PBX via: amportal restart
• Log into the PBX web portal, go to Settings > Music On Hold, and verify
  that (a) the new category is there and (b) the files you copied are listed.

Using the new category

The on-hold music category is specified in the Inbound Routes configuration.

Misc. Admin

Restarting Asterisk

# amportal restart

Changing the log level

First, ensure that the FreePBX Asterisk Logfiles module is installed.

• Within the FreePBX web UI, go to Admin > Module Admin
• Click the Check Online button
• Location the Asterisk Logfiles module and install it, if it isn’t
  already installed.

Next, adjust the logging.

• Go to Settings > Asterisk Logfile Settings

Backups

Needless to say, it’s a really good idea to backup your PBX. It’s
possible to screw things up beyond the point of recovery (especially using
experimental modules like Custom Contexts).

To schedule a routine backup, ensure that the Backup and Restore module is
installed. (By default, it should be.) Then, go to Admin > Backup. The
“Full Backup” template is a reasonable place to start.

Backups are written to /var/spool/asterisk/backup. Copying this directory
off to another machine is also a wise idea.

Troubleshooting

Can’t access voicemail

Password not accepted

You enter the password on the phone handset, but it isn’t accepted. Common
problem: The phone isn’t a SIP phone, and you’re using an adapter, but the
adapter’s DTMF method isn’t set to “SIP INFO” for that extension. Set the
DTMF method to “SIP INFO” and try again.

Feature Codes

These are the default settings. To change them, use the web admin UI and
go to Admin | Feature Codes.

References

• A Newcomer’s Guide to PBX in a Flash: http://nerdvittles.com/?p=755
• Incredible PBX Quick Start Guide: http://nerdvittles.com/?p=3026