Parts of a FreePBX System

There are several components that make up a FreePBX system. The main ones are outlined here.

Server

This is the system that runs FreePBX, Asterisk, and the rest of your telephony stack. Although it is possible to run this as a virtual machine, it is not recommended for production systems. Typically this is a physical machine sitting on the same LAN as your phones.

Phones

Obviously without phones, a PBX is not going to be able to do a whole lot. There are a lot of possibilities here: there are IP phones, analog phones, softphones. With FreePBX, you can even route calls to external numbers (eg, a cell phone).

1. IP Phones: IP phones are by far the best way to go when setting up a FreePBX-based system. They typically offer programmable buttons that make using various features easy (compared to dialing star codes, like *57). Most have multiple lines, which allow you to very easily manage more than one call to your extension. There are many manufacturers that make IP phones, and many many models that range in price from $80 up to $500. Within a single deployment, it's good to standardize on one line of products simply for end user ease-of-use, but of course you can mix and match all you like.
2. Analog Phones: Analog phones are the regular phones that you can plug into your phone line at home and just use. There are many ways to connect these to a FreePBX system. There are several manufacturers (such as Digium, Pika, Rhino, and Sangoma) that make PCI cards that will interface with Asterisk. These provide "fxs ports" which you can plug an analog phone into. Asterisk talks directly to these devices at the hardware level.

   Another way is to connect analog phones is with an ATA (analog telephone adapter), which is a network device. They have an ethernet port, and one or more fxs ports. They talk SIP or IAX2 over ethernet back to your asterisk system, and as far as Asterisk is concerned, they appear just like any other IP phone. The benefit to these devices is you can locate them closer to the phone, and just use your ethernet network instead of running analog wiring to the phone.

3. Softphones: This is software that you install on to your PC, that talks SIP or IAX2 back to Asterisk, and appears just like any other IP phone. You need a microphone and speakers of some sort (headset, usb device) to actually use it to place calls. There are many different softphones available for most platforms. They are useful for testing and connectivity while on the road (eg, on a laptop), but most people do not use them as a primary device because you're relying on a PC working, being turned on, user logged in, etc.
4. Legacy adapters: There are products on the market that allow you to connect existing legacy PBX phones to an Asterisk system, typically using SIP over ethernet.

PSTN Interface

Like phones, there are many ways to interface to the PSTN (public switched telephone network - the rest of the world). On the phone company side of things, there are many ways to provide interfaces:

1. POTS: Plain old telephone service. This is the type you have at home, it is an analog service and (in North America) is provided to you via an RJ11 jack. This supports one call (one channel) at a time, and is the most basic service. Generally each POTS line gets its own public phone number.
2. PRI: Primary Rate Interface. This is a digital line that carries 23 64kbps voice (B) channels, and one 64kbps signaling (D) channel used for call setup, caller id, etc. You can have zero up to hundreds of incoming DIDs ("direct inward dial" - an old telephony term that refers to a public phone number) on a PRI. It is the best interface to use when dealing with several lines.
3. ISDN BRI: Basic Rate Interface. This is a digital line carrying two 64kbps voice (B) channels and one 16kbps signaling (D) channel. It is not widely used with Asterisk systems (mostly due to the high price compared to POTS).
4. T1: Similar to a PRI line, it has 24 64kbps channels (though sometimes 56kbps), and signaling is done individually on each channel. There are many different configurations for T1s, and generally they are much harder to set up than a PRI.

There are many ways to connect all these interfaces to Asterisk:

1. PCI cards: There are several manufacturers (such as Digium, Pika, Rhino, and Sangoma) that make PCI cards that will interface with Asterisk. For analog lines, you need cards with fxo ports (note: most cards have the ability to have a mix of fxo and fxs ports). For digital lines, you can get cards with single ports, and cards with 4 ports (supporting a total of 4*23 = 92 channels, with PRI).
2. IP Gateways: These are similar to the ATA's described above, but have one or more fxo ports. You can also get gateways with digital ports.

VoIP PSTN Interface

Using a Voice over IP provider is another way to get connected to the PSTN. Most businesses do NOT use VoIP as their primary connection because it's unreliable compared to direct PSTN connections. It's susceptible to other internet traffic hogging bandwidth, and there are more points of failure (your ISP, your provider, your provider's ISP, and everyone else in between).

The provider will have PRI/T1 lines, and some kind of hardware and/or software that provides it to you as SIP or IAX2 (some providers even use Asterisk for this). There are all sorts of business models around this, but typically:

1. Flat-rate account: You pay a flat rate, and get some number of minutes that you are allowed to use. Usually with these accounts you get one DID (phone number) and can use one or two channels (simultaneous calls) at a time. Some of these providers lock you into using their hardware (they provide you a box that you can plug an analog phone into), so beware: you'll need some kind of fxo hardware to interface to this box (plus you lose quality by converting to analog in between, possibly introduce echo, etc). Having Asterisk talk SIP or IAX2 directly to the provider is always preferable.
2. Per-minute account: This is also sometimes marketed as "wholesale" VoIP. You pay per-minute, for all calls (local and long-distance, incoming and outgoing), usually around $0.01-$0.03/minute, depending on volume. Most providers will give you around four channels, but usually more if you ask (sometimes for a fee). You can get this service for origination (outbound) only, or you can get one or more DIDs for termination (incoming) calls. Typically you pay $3-5/month per DID, and most providers can get DIDs from anywhere in your country (or elsewhere).

Asterisk Extra Sounds

If you are using FreePBX in an embedded enviornment, you may be interested in reducing the overall size of the files necessary for FreePBX. FreePBX uses sounds from the asterisk-extra-sounds package (from Digium), but doesn't necessarily need all 34M of the sounds.

As of FreePBX 2.5, you need the following sounds from the asterisk-extra-sounds package to be in /var/lib/asterisk/sounds:

activated
added
all-circuits-busy-now
an-error-has-occured
at-tone-time-exactly
call-forwarding
call-fwd-no-ans
call-fwd-on-busy
call-fwd-unconditional
call-waiting
cancelled
cannot-complete-as-dialed
check-number-dial-again
day
de-activated
disabled
do-not-disturb
enabled
enter-conf-pin-number
enter-num-blacklist
enter-password
ent-target-attendant
extension
feature-not-avail-line
for
from-unknown-caller
goodbye
ha/phone
if-correct-press
im-sorry
info-about-last-call
is-curntly-unavail
is-in-use
is-set-to
is
location
number
num-was-successfully
one-moment-please
please-enter-your
pls-try-call-later
pm-invalid-option
press-0
press-1
press-2
press-3
press-4
press-5
press-6
press-7
press-8
press-9
press-star
privacy-to-blacklist-last-caller
reception
sorry-youre-having-problems
speed-dial-empty
speed-dial
telephone-number
to-call-this-number
to-listen-to-it
with
your

Install Procedure for Centos 4.3

Centos 4.3 Installation Walkthrough

CentOS 4.3 (CentOS) is the distribution used throughout this guide.
We believe that the goals of the distribution are in good alignment
with the mission-critical nature of a corporate telephone system.
CentOS ISOs can be downloaded from a number of mirror sites. Check the
official CentOS website for more information.

Detailing a Linux installation is beyond the scope of this
document. There are numerous articles, HOW-TOs and books available to
the individual that deal with this subject. Therefore, for the purposes
of this document it is assumed that the CentOS installation is that of
a Server system. Furthermore, for the purposes of this
document it is assumed that the partitioning of the hard disk drive was
done automatically by selecting Autopartition when prompted, and that no previous partitions existed on the drive prior to installation.

Important Installation Notice During the installation you will be prompted about Firewalls and Selinux. Both of these MUST BE DISABLED.
The Sections to disable are highlighted in red below. After clicking
next, you will be prompted if you are sure this is correct - Click on Proceed.

Package Group Selection
Whilst It is not recommended to use the X Window System on a
production freePBX server, it is possible. If you're only doing this
for a test, or experementation, feel free to install X and Gnome or
KDE. If you are planning on using this as a production system, please
avoid installing X unless it's absoloutely necessary.

freePBX has several requirements (which we will cover in a later
section) but at this point of the CentOS installation ensure that at
least the following package groups are selected

• Web Server
• Mail Server (Not selected by default)
• MySQL Database (Not selected by default)
• Development Tools (Not selected by default)

After you've done this, the machine will install CentOS, install,
and reboot. At this stage, you have a functioning Linux system!

Post-Install Configuration

After your machine reboots, you need to log in as 'root' - You were
prompted for the root password on installation. When you log in
successfully, you will have a prompt li

For performance and security reasons it is important to update the system immediately after install. CentOS uses yum (or up2date but that is not a recommended way of doing updates) for this purpose. In this document we will use yum:

[root@dhcp1 ~]# yum -y update

Setting up Update Process

Setting up repositories

...etc....

...etc...

Update: gnupg.i386 0:1.2.6-3 python.i386 0:2.3.4-14.2 sendmail.i386 0:8.13.1-3.RHEL4.3 tzdata.noarch 0:2006a-2.EL4

Complete!

root[@dhcp1 ~]#

Additional Package Installation to Satisfy freePBX dependencies
You can check if a particular package is installed by doing either:

[root@dhcp1 ~]# yum info [package]

or:

[root@dhcp1 ~]# rpm -qa | grep [package]

If the package is not installed, install it by using yum:

[root@dhcp1 ~]# yum install [package]

Full documentation on 'yum' is available by typing man yum.

The following packages need to be additionally installed with yum:

[root@dhcp1 ~]# yum install gcc
libxml2-devel libtiff-devel mysql-server php-gd php-mysql kernel-devel
kernel-smp-devel bison ncurses-devel audiofile-devel subversion
libogg-devel openssl-devel mysql-devel

lame is not available through a yum repository; but it can be obtained and installed from Dag Wieers' RPM repository:

[root@dhcp1 ~]# rpm -ivh http://apt.sw.be/redhat/el4/en/i386/RPMS.dag/lame-3.96.1-2.2.el4.rf.i386.rpm

Satisfying freePBX's PERL module dependencies
freePBX, from version 2.1 does not have any specific perl
dependancies. There used to be a big list here, but we finally managed
to get rid of all of them!

Get the latest freePBX files
You may wish to check that the link specified here is actually the latest and greatest. Look at the files available on Source Forge and pick the latest one there.

[root@dhcp1 ~]# cd /usr/src

[root@dhcp1 src]# wget http://easynews.dl.sourceforge.net/sourceforge/amportal/freepbx-2.1.1.tar.gz

[root@dhcp1s src]# tar zxf freepbx-2.1.1.tar.gz

A pause while the files are extracted...

[root@dhcp1 src]#

Getting all the required Asterisk and Zaptel files.

[root@dhcp1 ~]# cd /usr/src

[root@dhcp1 src]# svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk

..Lots of files...

[root@dhcp1 src]# svn co http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons

..Lots of files...

[root@dhcp1 src]# svn co http://svn.digium.com/svn/asterisk-sounds/trunk asterisk-sounds

..Lots of files...

[root@dhcp1 src]# svn co http://svn.digium.com/svn/zaptel/branches/1.2 zaptel

..Lots of files...

[root@dhcp1 src]# svn co http://svn.digium.com/svn/libpri/branches/1.2 libpri

..Lots of files...

Patch and Compile zaptel (and libpri)
If you plan on useing IAX or conferencing and _don't_ have any digium hardware skip this part and read this ztdummy install guide then continue on at "Compile Asterisk"

[root@dhcp1 src]# cd /usr/src/zaptel

[root@dhcp1 zaptel]# cp ztdummy.c ztdummy.c.orig

[root@dhcp1 zaptel]# sed -i "s/if 0/if 1/" ztdummy.c

[root@dhcp1 zaptel]# make

If you get an error that looks like this:

/usr/src/zaptel/zaptel.c:420: error: syntax error before "zone_lock"

/usr/src/zaptel/zaptel.c:420: warning: type defaults to `int' in declaration of `zone_lock'

/usr/src/zaptel/zaptel.c:420: error: incompatible types in initialization

..10 or so more lines..

it means that your CentOS header files have an error in them.

This is a known bug and is easily repaired by

[root@dhcp1 zaptel]# sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h

You can now retry the make command. After it's finished, you need to run make install and make config. If you will be using a Digium or Sangoma telephony card that supports T1/E1 signaling do this step as well:

[root@dhcp1 zaptel]# cd /usr/src/libpri

[root@dhcp1 libpri]# make install

Compile Asterisk
Now you get to compile the centre of the package - asterisk! This is relatively paineless:

[root@dhcp1 libpri]# cd /usr/src/asterisk

[root@dhcp1 asterisk]# mkdir /var/run/asterisk

[root@dhcp1 asterisk]# make install

[root@dhcp1 asterisk]# make config

Take a couple of minutes now and configure your zaptel
files, before continuing. If you think you'll want to have Asterisk and
freePBX itself handle faxing (rather than using a dedicated fax
device), you should read the Faxing page.

Create user and set permissions
Unfortunately, issues in Asterisk 1.2 require us
to run the web server process as the same user as asterisk. In this
situation, it's easier for us to run httpd as 'asterisk', rather than
asterisk as 'httpd', as there's far less configuration that needs to be
done.

[root@dhcp1 ~l]# useradd -c "Asterisk PBX" -d /var/lib/asterisk asterisk

[root@dhcp1 ~]# chown asterisk /var/lib/php/session/

Using nano (or your favourite editor, but nano is fine), you need to change User apache and Group apache to User asterisk and Group asterisk.

[root@dhcp1 ~]# nano +227 /etc/httpd/conf/httpd.conf (Push Control-X to save when you've finished)

You also want to change AllowOverride None to AllowOverride All

[root@dhcp1 ~]# nano +311 /etc/httpd/conf/httpd.conf (Push Control-X to save when you've finished)

Set up MySQL
Before you can do anything to MySQL, you need to make sure it's running:

[root@dhcp1 ~]# /etc/init.d/mysqld start

Initializing MySQL database: [ OK ]

Starting MySQL: [ OK ]

[root@dhcp1 ~]#

Now, to configure the databases for freePBX:

[root@dhcp1 ~]# cd /usr/src/freepbx-2.1.1

[root@dhcp1 freepbx-2.1.1]# mysqladmin create asterisk

[root@dhcp1 freepbx-2.1.1]# mysqladmin create asteriskcdrdb

[root@dhcp1 freepbx-2.1.1]# mysql asterisk < SQL/newinstall.sql

[root@dhcp1 freepbx-2.1.1]# mysql asteriskcdrdb < SQL/cdr_mysql_table.sql

They also need to be secured, so that not just anyone can access
them. freePBX will prompt you for a database password when you do the
install. You need to pick that now. We'll assume that you've picked
'asteriskuser' and 'amp109' - you probably shouldn't use these, as they
are well known passwords for Asterisk@Home builds. If anyone's trying
to attack your machine, they will try this.

[root@dhcp1 freepbx-2.1.1]# mysql

Welcome to the MySQL monitor. Commands end with ; or \g.

Your MySQL connection id is 8 to server version: 4.1.16

Type 'help;' or '\h' for help. Type '\c' to clear the buffer.

mysql> GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY 'amp109';

Query OK, 0 rows affected (0.00 sec)

mysql> GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY 'amp109';

Query OK, 0 rows affected (0.00 sec)

mysql> flush privileges;

Query OK, 0 rows affected (0.00 sec)

mysql> \q

Bye

[root@dhcp1 freepbx-2.0.1]

Now, after all of this, you need to pick a root 'mysql' password.
For this, we'll pretend it's 's33kret'. If you need to do anything else
with mysql, you'll need to provide this password.

[root@dhcp1 freepbx-2.1.1]# mysqladmin -u root password 's33kret'

Build the cdr_mysql module for Asterisk (Yep, more compiling!)

[root@dhcp1 freepbx-2.1.1]# cd /usr/src/asterisk-addons

[root@dhcp1 freepbx-2.1.1]# cp Makefile Makefile.orig

[root@dhcp1 freepbx-2.1.1]# sed -i 's/SOURCE/SOURCE -DMYSQL_LOGUNIQUEID/' Makefile

[root@dhcp1 freepbx-2.1.1]# make && make install

Install freePBX at last!
You're there - you've done the hard yards, and finally you can install freePBX!

WARNING! If you have an existing Asterisk installation, the script below will overwrite your Asterisk configuration files. Backup your

/etc/asterisk directory before running.

$cd /usr/src/freepbx-2.1-beta1

$./install_amp

Checking for PEAR DB..OK

Checking for PEAR Console::Getopt..OK

Checking for libasteriskperl (perl bindings for asterisk)...Checking user..OK

Checking for /etc/amportal.conf../etc/amportal.conf does not exist, copying default

Creating new /etc/amportal.conf

Enter your USERNAME to connect to the 'asterisk' database: [asteriskuser]

Enter your PASSWORD to connect to the 'asterisk' database: [amp109]

Enter the hostname of the 'asterisk' database: [localhost]

Enter a USERNAME to connect to the Asterisk Manager interface: [admin]

Enter a PASSWORD to connect to the Asterisk Manager interface:[amp111]

Enter the path to use for your AMP web root:[/var/www/html]

Enter the path to use for your FOP web root:[/var/www/html/panel]

Created /var/www/html/panel

Enter the path to your Apache cgi-bin:[/var/www/cgi-bin]

Enter the IP ADDRESS or hostname used to access the AMP web-admin:[xx.xx.xx.xx] The IP Address of your Asterisk Machine

Enter a PASSWORD to perform call transfers with the Flash Operator Panel: [passw0rd]

Use simple Extensions [extensions] admin or separate Devices and Users [deviceanduser]? extensions

Enter directory in which to store AMP executable scripts: [/var/lib/asterisk/bin]

Created /var/lib/asterisk/bin

Enter directory in which to store super-user scripts: [/usr/sbin]

/etc/amportal.conf writtenOK

Reading /etc/amportal.conf..OK

Checking for /etc/asterisk/asterisk.conf../etc/asterisk/asterisk.conf does not exist, copying default

OK

Reading /etc/asterisk/asterisk.conf..OK

Connecting to database..OK

Checking current version of AMP..1.10.010beta1

Installing new AMP files..OK

Configuring install for your environment..OK

Setting permissions on files..OK

Checking for upgrades..5 found

Upgrading to 1.10.010..

Upgrading to 1.10.010..OK

Upgrading to 2.0beta1..

-> Running PHP script /usr/src/freepbx-2.0-beta4/upgrades/2.0beta1/emergencycid.php

-> Running SQL script /usr/src/freepbx-2.0-beta4/upgrades/2.0beta1/tables.sql

PHP Notice: Undefined variable: data in /usr/src/freepbx-2.0-beta4/install_amp on line 305

Upgrading to 2.0beta1..OK

Upgrading to 2.0beta2..

Upgrading to 2.0beta2..OK

Upgrading to 2.0beta3..

-> Running PHP script /usr/src/freepbx-2.0-beta4/upgrades/2.0beta3/fixgotovm.php

Updating existing voicemail destinations..

..OK

Upgrading to 2.0beta3..OK

Upgrading to 2.0beta4..

Upgrading to 2.0beta4..OK

Generating AMP configs..

Generating Configurations.conf..

Checking for PEAR DB..OK

Checking for PEAR Console::Getopt..OK

Checking for /etc/amportal.conf..OK

Reading /etc/amportal.conf..OK

Connecting to database..OK

Please Reload Asterisk by visiting http://XXX.XXX.XXX.XX/admin

Generating AMP configs..OK

Restarting Flash Operator Panel..-bash: /var/www/html/admin/bounce_op.sh: Permission denied

OK

Please Reload Asterisk by visiting http://XXX.XXX.XXX.XX/admin

If you get any warnings or errors in the last part of the output,
they're usually not traumatic, but please use the IRC Support tool to
report a bug to the developers.

amportal control script

Starting with version 1.10.004, freePBX provided a new control
script. The functionality of which is to start, stop or kill services
in the freePBX environment, or to set permissions on directories/files
in the freePBX environment:

$amportal

----------AMP Control Script-----------

Usage: amportal start|stop|kill|chown

start: Starts Asterisk and Flash Operator Panel server

stop: Gracefully stops Asterisk and the FOP server

restart: Stop and Starts

kill: Kills Asterisk and the FOP server

chown: Sets appropriate permissions on files

The amportal script is the recommended way to stop and start asterisk:

$ /usr/sbin/amportal stop

$ /usr/sbin/amportal start

19.Automatic start-up

echo /usr/sbin/amportal start >> /etc/rc.local

Ensure services are starting at boot time and reboot

In order to access and use freePBX we will want both Apache (httpd)
and MySQL (mysqld) to be started at boot. You can check to see if they
are setup to start at boot by using chkconfig:

[root@dhcp1 freepbx-2.1.1]# chkconfig --list httpd

httpd 0:off 1:off 2:off 3:off 4:off 5:off 6:off

[root@dhcp1 freepbx-2.1.1]# chkconfig --list mysqld

mysqld 0:off 1:off 2:off 3:off 4:off 5:off 6:off

Here we see that both httpd and mysqld have off
across the board (runlevels). chkconfig can also be used to turn on a
particular service, which you would want to do in this case.

^[root@dhcp1 freepbx-2.1.1]# chkconfig httpd on

[root@dhcp1 freepbx-2.1.1]# chkconfig mysqld on

You can now access freePBX with your web browser.

The first time you click on the FreePBX Administration link you
will be prompted for a username and password. Use admin and admin.
CREATE A NEW ADMINISTRATIVE USER IMMEDIATELY AFTER LOGIN.

Install process for CentOS 5.1

Install process for CentOS 5.1

1. Install CentOS, enabling the following packages:

*DNS Server

*Web Server

*Mail Server

*MySQL Database

*Development Tools

yum install nano

reboot

2. Edit Network settings

nano /etc/sysconfig/network

HOSTNAME=internal.hostname.DOMAIN.com (Set your internal hostname name here)

Ctrl-X to save, 'Y' to confirm

nano /etc/sysconfig/network-scripts/ifcfg-eth0

IPADDR=192.168.1.20

NETMASK=255.255.255.0

GATEWAY=192.168.1.1

NETWORK=192.168.1.0

Ctrl-X to save, 'Y' to confirm

echo "options {" >> /etc/named.conf

echo " directory \"/var/named\";" >> /etc/named.conf

echo " dump-file \"/var/named/data/cache_dump.db\";" >> /etc/named.conf

echo " statistics-file \"/var/named/data/named_stats.txt\";" >> /etc/named.conf

echo "};" >> /etc/named.conf

echo "include \"/etc/rndc.key\";" >> /etc/named.conf

service named start

chkconfig named on

nano /etc/resolv.conf

search internal.DOMAIN.com (Set your internal domain name here)

nameserver 192.168.1.5

nameserver 127.0.0.1

nano /etc/hosts

127.0.0.1 internal.hostname.DOMAIN.com (Set your internal hostname name here)

127.0.0.1 asterisk1.local

127.0.0.1 localhost

Ctrl-X to save, 'Y' to confirm

iptables -P INPUT ACCEPT

iptables -P OUTPUT ACCEPT

iptables -P FORWARD ACCEPT

iptables -F

iptables -X

/etc/init.d/iptables save

service network restart

3. Update:

yum -y update

4. Disable Selinux:

echo "selinux=disabled" > /etc/selinux/config

reboot

5. Install dependencies and extra packages:

yum install e2fsprogs-devel keyutils-libs-devel krb5-devel libogg libselinux-devel libsepol-devel libxml2-devel libtiff-devel gmp php-pear php-pear-DB php-gd php-mysql php-pdo kernel-devel ncurses-devel audiofile-devel libogg-devel openssl-devel mysql-devel zlib-devel perl-DateManip sendmail-cf sox

cd /usr/src

wget http://easynews.dl.sourceforge.net/sourceforge/lame/lame-3.97.tar.gz

tar zxvf lame-3.97.tar.gz

cd lame-3.97

./configure

make

make install

6. Install Asterisk and FreePBX:

cd /usr/src

wget http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz

wget http://downloads.digium.com/pub/asterisk/asterisk-addons-1.4-current.tar...

wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-l...

wget http://downloads.digium.com/pub/libpri/libpri-1.4-current.tar.gz

wget http://mirror.freepbx.org/freepbx-2.6.0.tar.gz

tar zxvf asterisk-1.4-current.tar.gz

tar zxvf asterisk-addons-1.4-current.tar.gz

tar zxvf dahdi-linux-complete-current.tar.gz

tar zxvf libpri-1.4-current.tar.gz

tar zxvf freepbx-2.6.0.tar.gz

cd /var/lib/asterisk/sounds

wget http://downloads.digium.com/pub/telephony/sounds/asterisk-extra-sounds-e...

tar zxvf asterisk-extra-sounds-en-gsm-current.tar.gz

cd /usr/src/dahdi-linux-complete-CURRENT

make

make install

make config

/sbin/ztcfg

echo "/sbin/ztcfg" >> /etc/rc.d/rc.local

cd /usr/src/libpri-1.4-CURRENT

make clean

make

make install

cd /usr/src/asterisk-1.4-CURRENT

useradd -c "Asterisk PBX" -d /var/lib/asterisk asterisk

mkdir /var/run/asterisk

mkdir /var/log/asterisk

chown -R asterisk:asterisk /var/run/asterisk

chown -R asterisk:asterisk /var/log/asterisk

chown -R asterisk:asterisk /var/lib/php/session/

nano +231 /etc/httpd/conf/httpd.conf

Change User apache and Group apache to User asterisk and Group asterisk.

Ctrl-X to save, 'Y' to confirm

nano +329 /etc/httpd/conf/httpd.conf

Change AllowOverride None to AllowOverride All

Ctrl-X to save, 'Y' to confirm

./configure

make

make install

/etc/init.d/mysqld start

cd /usr/src/freepbx-2.6.0

mysqladmin create asterisk

mysqladmin create asteriskcdrdb

mysql asterisk < SQL/newinstall.sql

mysql asteriskcdrdb < SQL/cdr_mysql_table.sql

mysql

GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY 'SOMEPASSWORD';

GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY 'SOMEPASSWORD';

flush privileges;

\q

mysqladmin -u root password 'SOMEPASSWORD'

cd /usr/src/asterisk-addons

./configure

make

make install

cd /usr/src/freepbx-2.6.0

./start_asterisk start

yum install php-pear-DB

yum install php-mysql

./install_amp --username=asteriskuser --password=SOMEPASSWORD

echo "/usr/local/sbin/amportal start" >> /etc/rc.local

chkconfig httpd on

chkconfig mysqld on

Open browser to http://ipaddressofpbx/admin

Click red bar in FreePBX

7. Fix ARI password:

nano -w /var/www/html/recordings/includes/main.conf.php

$ari_admin_password = "SOMEPASSWORD";

Ctrl-X to save, 'Y' to confirm

8. Configure Sendmail:

nano /etc/mail/sendmail.mc

define(`SMART_HOST', `relay.DOMAIN.com)dnl

MASQUERADE_AS(`pbx.DOMAIN.com')dnl

FEATURE(`masquerade_envelope')dnl

Ctrl-X to save, 'Y' to confirm

make -C /etc/mail

9. Edit sip_nat.conf for proper NAT:

nano /etc/asterisk/sip_nat.conf

localnet=192.168.1.0/255.255.255.0

externhost=pbx.DOMAIN.com (Set your external hostname name here)

externrefresh=10

fromdomain=DOMAIN.com (Set your external domain name here)

nat=yes

qualify=yes

canreinvite=no

Ctrl-X to save, 'Y' to confirm

10. Add extra codecs to config:

nano /etc/asterisk/sip_custom.conf

allow=gsm

allow=h261

allow=h263

allow=h263p

videosupport=yes

Ctrl-X to save, 'Y' to confirm

nano /etc/asterisk/iax_custom.conf

allow=gsm

allow=h261

allow=h263

allow=h263p

videosupport=yes

Ctrl-X to save, 'Y' to confirm

asterisk -rx reload

11. Edit voicemail config:

nano /etc/amportal.conf

If the web interface on your PBX will be accessible on the internet:

AMPWEBADDRESS=pbx.DOMAIN.com (Set your external hostname name here)

If the web interface on your PBX will be accessible only on your internal network:

AMPWEBADDRESS=internal.hostname.DOMAIN.com (Set your internal hostname name here)

Ctrl-X to save, 'Y' to confirm

or if your users will NOT have access to the web interface:

nano /etc/asterisk/vm_email.inc

remove "Visit http://AMPWEBADDRESS/cgi-bin/vmail.cgi?action=login&mailbox=${VM_MAILBOX} to check your voicemail with a web browser.\n"

Ctrl-X to save, 'Y' to confirm

nano /etc/asterisk/vm_general.inc

serveremail=pbx@DOMAIN.com ; Who the e-mail notification should appear to come from

fromstring=DOMAIN PBX ; Real name of email sender

Ctrl-X to save, 'Y' to confirm

12. Fix MOH directory:

ln -s /var/lib/asterisk/moh /var/lib/asterisk/mohmp3

asterisk -rx reload

14. Open browser to http://ipaddressofpbx

Done!

Optional Package:

ConfControl is a partial rewrite of Web-MeetMe which includes only the conference control page and not all the scheduling options. It includes a replacement index.html page which adds a link to ConfControl from the FreePBX home page.

cd /var/www/html

wget http://www.2l2o.com/asterisk/ConfControl1.0.tar.gz

tar zxvf ConfControl1.0.tar.gz

rm ConfControl1.0.tar.gz

Install Process for ClarkConnect

Detailed guides for installing FreePBX on ClakConnect can be found here:

http://samyantoun.50webs.com/asterisk/freepbx/clarkconnect/

Install Process for Debian

A good guide is http://www.squishychicken.com/index.php?option=com_content&task=view&id=13&Itemid=2. It's for AMP and Asterisk 1.2, but can used to install freePBX instead.

Updated for FreePBX and other software version changes. I also felt it needed to be made more idiot proof for dummies like me who spent many frustrating hours trying to get it to work.
http://powerontech.com/freepbx-on-debian.htm

There is a good Ubuntu guide here that can be of assistance, until somebody writes an updated guide for freePBX 2.3 and Debian Etch.

Install Process for freeBSD

0.- In the meantime freepbx is not in the oficial freebs ddistribution download the port from here

http://www.freepbx.org/attachment/ticket/761/freepbx-2.1.tar

1.- Choose what apache, mysql and php version you want to use. For example, you could put this in your make.conf:

  DEFAULT_PHP_VER=5

  DEFAULT_MYSQL_VER=50

  APACHE_PORT=www/apache20

2.- cd /usr/ports/misc/freepbx

3.- If you have php previously install make sure you have pear in your php.ini include paths.

3.- make install

4.- drink a coffee

5.- Read the post instalation notes:

       1) enable .php files in your apache config

       2) adding index.php as default index files

       3) add pear and /.../admin to your php.ini includes

       4) Make sure asterisk and mysql are running fine

       5) Make sure asterisk accepts connections (manager.conf)

6.- http://localhost/FreePBX

7.- Complete this document!

Install Process for Gentoo

This is a work in progress

Notes
Basic gentoo knowledge is required, if you don't know what a USE flag then this will be hard to follow.

Look the the requirements in the INSTALL file and get your apache/php flags correct and reemerge dev-lang/php if needed.

emerge asterisk >= 1.2 - http://gentoo-portage.com/net-misc/asterisk

and asterisk-addons - http://gentoo-portage.com/net-misc/asterisk-addons

note: as of this writing the correct ebuild is asterisk-1.2.9_p1

Both are masked with ~x86 so (if you're not on x86 modify as needed) umask them with

`echo "net-misc/asterisk ~x86" >> /etc/portage/package.keywords`

and

`echo "net-misc/asterisk-addons ~x86" >> /etc/portage/package.keywords`

To emerge:

`emerge asterisk asterisk-addons`

Note: I have `net-misc/asterisk zaptel speex mysql -vmdbmysql` in
my package.use for asterisk but you may need otherwise, read the use
flags descriptions at gentoo-portage.com

This will get asterisk to be ready to futher configure.

Amportal Issues:

• It appears some gentoo paths are not compatible with amportal:

in /usr/sbin/safe_asterisk change

ASTSBINDIR=/sbin to

ASTSBINDIR=/usr/sbin

• amportal can't run the safe_opserver because the asterisk user on my machine had /bin/false for it's shell

check in /etc/passwd and make the asterisk user have /bin/bash as the shell if it's not already such.

I also needed asterisk ownership on the htdocs folder above the freepbx files.

• note: this article used to tell you how to implement it
  with lighthttpd but I've cleared out that info as it is more specific
  than it needs to be and novices may make serious mistakes following
  them.

Install Process for Poundkey

"Engineered by Digium in conjunction with rPath, Pound Key includes all
the Linux components necessary to run, debug and build Asterisk, and
only those components. You no longer have to worry about kernel
versions and package dependencies. Unlike other Linux distributions
used to deploy Asterisk, no unnecessary components that might
compromise security or performance are included."

While technically true, PoundKey is missing several components
needed for FreePBX to run. Below are most of the commands neded to
ensure a smooth installation of FreePBX when following the normal
INSTALL document.

Install missing rPath components from the primary distributions:

conary update libxml2 libtiff bison audiofile php-mysql m4

Install lame from the Media Center Linux repository:

conary update lame=steel.rpath.org@rpl:devel

Repair PEAR for proper operation of the install_amp script:

pear install DB

Install perl MIME::Types:

perl -MCPAN -e "install MIME::Types"

Once these steps are complete, the normall INSTALL document can be easily followed.

Install Process for SuSE

FreePBX 2 has been tested with SuSe LES 9 and SuSe 10.

In reality seems fairly solid with not a lot of work needing to be
done to run FreePBX. Mainly check that you have DB (installable via
PEAR if required) and PHP4-GETTEXT (available as part of the SUSE
install) installed. If you have any PHP problems check that you have
the line include_path = ".:/usr/share/php" in /etc/php.ini. (The '.:' is the important bit)

To install DB, or if you aren't sure whether you have it installed, you can use pear. pear list will show a list of installed packages. To install DB either run pear install DB
on the command line or visit http://pear.php.net/package/DB and
download the latest stable version, 1.7.6 at present. A pear install
should download it and install it but it will complain about any
dependancies it needs and had a tendancy to fail on my system due for
this exact reason. The best way I found was to download the file onto
my SuSe? system and run pear install -n <location/filename>
manually. This tells pear to ignore any dependancies, as otherwise it
will probably complain about pear itself! Once finished you should have
DB.php and a DB directory in /usr/share/php.

When installing FreePBX remember that the webserver root is
different to the one defaulted to in the install_amp script. Your
webserver root should be /srv/www/htdocs so use that as the root and a
bit of common sense to change the panel and cgi-bin paths.

No other file changes should be required to get FreePBX working.

Install Process for Ubuntu 6.06

In this document there are various boxes, with text inside them.
These are examples of what you see on your screen, what you should
type, and/or the expected responses. For example:

rob@rob-laptop:~$ id

uid=1000(rob) gid=1000(rob)
groups=4(adm),20(dialout),24(cdrom),25(floppy),
29(audio),30(dip),44(video),46(plugdev),
106(lpadmin),110(scanner),112(admin),1000(rob)

In that situation, you would type 'id' and the response would be
similar to the response indicated. If you see an error, that probably
means you've typed something incorrectly, but it could also suggest
other problems. Read the error carefully, and if you don't know how to
fix it, feel free to post to the forums or ask on IRC.

Operating System Installation
Installation of Ubuntu is out of the scope of this document.. As a
reference point this document was written with accepting the defaults
for everything suggested in the installer..

Post Installation Configuration
After your machine has rebooted and you've logged in, you need to
switch to the 'root' account. Whilst installing you were prompted for a
password, which is what you must enter when prompted below:

rob@rob-laptop:~$ sudo su -

Password: Enter Password Here

root@rob-laptop:~#

This gives you full control of the system. It's the equivlent of typing 'sudo' before every command.

You now need to ensure that your machine is able to access the
internet, and once that is working you can proceed with the
installation Usually being able to browse the internet with Firefox is
a good indication that you won't be having any problems.

Check for updates and install required packages
Before running apt-get, you must edit the sources.list file so that you can install from the "universe".

nano +17 -w /etc/apt/sources.list

Remove the # signs from this line and add multiverse at the end:

deb http://us.archive.ubuntu.com/ubuntu/ dapper universe multiverse

Multiverse gives you access to a lot more packages then the standard repository does, useful for future reference.

Unless you've added extra repositories, your sources.list should look something like:

deb http://archive.ubuntu.com/ubuntu/ dapper main restricted universe multiverse

deb http://archive.ubuntu.com/ubuntu/ dapper-updates main restricted universe multiverse

deb http://archive.ubuntu.com/ubuntu/ dapper-backports main restricted universe multiverse

deb http://security.ubuntu.com/ubuntu/ dapper-security main restricted universe multiverse

In addition to this, you must update apt's package lists"

apt-get update

You need to ensure that your machine is up to date with the current
security packages release by Ubuntu. After doing so, an apt-get of the
modules below will install all the requirements for freePBX

root@rob-laptop:~# apt-get install
php5 php5-cli php5-mysql mysql-server php-pear php-db openssh-server
curl sox apache2 subversion build-essential libncurses5-dev libssl-dev
linux-headers-`uname -r` libmysqlclient15-dev

... Please copy-and-paste that line, rather than trying to type it in.

Reading package lists... Done

Building dependency tree... Done

The following extra packages will be installed:

apache-common apache2-common apache2-mpm-prefork apache2-utils

... Several more lines of automatically imported packages ...

mysql-server-5.0 openssh-server php-db php-http php-mail php-net-smtp

php-net-socket php-pear php-xml-parser php4 php4-cli php4-common php4-mysql

php4-pear php5-common sox ssl-cert zlib1g-dev

0 upgraded, 39 newly installed, 0 to remove and 6 not upgraded.

Need to get 40.6MB of archives.

After unpacking 107MB of additional disk space will be used.

Do you want to continue [Y/n]? y

Get:1 http://au.archive.ubuntu.com dapper/main libpcre3 6.4-1.1ubuntu4 [174kB]

Get:2 http://security.ubuntu.com dapper-security/main libapr0 2.0.55-4ubuntu2.1 [132kB]

Get:3 http://au.archive.ubuntu.com dapper/main ssl-cert 1.0.13 [9526B]

Get:4 http://au.archive.ubuntu.com dapper/main curl 7.15.1-1ubuntu2 [168kB]

... The machine now proceeds to download and install packages ...

Setting up sox (12.17.9-1) ...

root@rob-laptop:~#

Downloading and Installing Asterisk

Previous comment:
Options, if you use edgy instead of dapper, you will get the
latest version of asterisk without needing to compile etc etc etc, all
you need to do is: apt-get install asterisk and skip to the section on
MySQL... Dapper also has packaged versions of asterisk and is quite
useable also, and is a lot easier to manage then compiling and
re-compiling to upgrade all the time

Rob's Response:
Well, I tried this, (on 6.06 LTS) and got Asterisk-1.2.7.1, and
Zaptel-1.2.5. Both of these are _woefully_ out of date. Asterisk has 3
Denial-Of-Service bugs and 2 security bugs, and Zaptel doesn't have the
proper echo cancellation enabled in it. Yes. Maybe it might be easier
to type in 'apt-get install asterisk zaptel', but it'll be crap.
Compile from source, it's not that hard. To make it easier, I've put
one box at the bottom of the downloading section that you can
copy-and-paste from to install everything from source.

Downloading
We will be using subversion to download the latest version of the
1.2 branch of Asterisk, Zaptel, LibPRI and Asterisk-Addons.
asterisk-sounds will be the latest version.

root@rob-laptop:/usr/src# svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2

... Lots of files are downloaded ...

root@rob-laptop:/usr/src# svn co http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2

... Zaptel files download ...

root@rob-laptop:/usr/src# svn co http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2

... LibPRI downloads - Note, this is quite small, only about 15 files. This is normal ...

root@rob-laptop:/usr/src# svn co http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2

... Asterisk-Addons downloads...

root@rob-laptop:/usr/src# svn co http://svn.digium.com/svn/asterisk/trunk/sounds asterisk-sounds

... The default Sounds package downloads now. This can be quite large ...

root@rob-laptop:/usr/src#

Compiling and Installing
You now need to compile and install the latest version of asterisk.

root@rob-laptop:/usr/src# cd libpri-1.2

root@rob-laptop:/usr/src/libpri-1.2# make install

gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o copy_string.o copy_string.c

gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -c -o pri.o pri.c

... libpri compiles ...

install -m 644 libpri.a /usr/lib

if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi

root@rob-laptop:/usr/src/libpri-1.2# cd ../zaptel-1.2

root@rob-laptop:/usr/src/zaptel-1.2# make install config

cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DHOTPLUG_FIRMWARE -c -o
gendigits.o gendigits.c

cc -o gendigits gendigits.o -lm

./gendigits > tones.h

... zaptel compiles - Note that any errors here are usually
because of you not having the correct version of linux-headers'
installed ...

ZAPTELVERSION="SVN-branch-1.2-r1468" build_tools/make_version_h > version.h.tmp

if cmp -s version.h.tmp version.h ; then echo; else \

mv version.h.tmp version.h ; \

fi

... Zaptel compiles ...

root@rob-laptop:/usr/src/zaptel-1.2# cd ../asterisk-1.2

root@rob-laptop:/usr/src/asterisk-1.2# make install

if cmp -s .cleancount .lastclean ; then echo ; else \

make clean; cp -f .cleancount .lastclean;\

fi

make1: Entering directory `/usr/src/asterisk-1.2'

... Asterisk Compiles ...

root@rob-laptop:/usr/src/asterisk-1.2# cd ../asterisk-addons-1.2

root@rob-laptop:/usr/src/asterisk-addons-1.2# make install

./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`

make -C format_mp3 all

... Asterisk-Addons now install ...

root@rob-laptop:/usr/src/asterisk-addons-1.2# cd ../asterisk-sounds/

root@rob-laptop:/usr/src/asterisk-sounds# make install

... Lots of additional sound files are installed here ...

root@rob-laptop:/usr/src/asterisk-sounds#

Copy-And-Paste this, for ease of installation:

cd /usr/src
svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
svn co http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
svn co http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2
svn co http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2
svn co http://svn.digium.com/svn/asterisk/trunk/sounds asterisk-sounds
cd /usr/src/libpri-1.2 && make install
cd /usr/src/zaptel-1.2
sed -i 's!^#define ECHO_CAN_KB1!/* #define ECHO_CAN_KB1 */!' zconfig.h
sed -i 's!/\* #define ECHO_CAN_MG2 \*/!#define ECHO_CAN_MG2!' zconfig.h
make install
cd /usr/src/asterisk-1.2 && make install
cd /usr/src/asterisk-addons-1.2
sed -i 's/_GNU_SOURCE/_GNU_SOURCE -DMYSQL_LOGUNIQUEID/' Makefile
make install

If all you wanted to do was install Asterisk on a Ubuntu machine,
you're done - you now have a fully functional Asterisk box, for you to
play with as you wish. If you want, you can run 'make samples' in the
asterisk-1.2 directory to install some example configuration files for
you to play with. However, since you're reading this on the FreePBX
site, we're now up to isetting the machine up and nstalling FreePBX.

Create user and set permissions
Unfortunately, issues in Asterisk 1.2 require us
to run the web server process as the same user as asterisk. In this
situation, it's easier for us to run httpd as 'asterisk', rather than
asterisk as 'httpd', as there's far less configuration that needs to be
done.

root@rob-laptop:~# addgroup asterisk

Adding group `asterisk' (1001)...

Done.

root@rob-laptop:~# useradd -g asterisk -c "Asterisk PBX" -d /var/lib/asterisk asterisk

root@rob-laptop:~# mkdir /var/run/asterisk

root@rob-laptop:~# chown -R asterisk /var/lib/php5

Using nano (or your favourite editor, but nano is fine), you need to change User apache and Group apache to User asterisk and Group asterisk.

[root@dhcp1 ~]# nano +101 /etc/apache2/apache2.conf (Push Control-X to save when you've finished)

You also want to change AllowOverride None to AllowOverride All

[root@dhcp1 ~]# nano +12 /etc/apache2/sites-enabled/000-default (Push Control-X to save when you've finished)

And then restart asterisk to re-load its configuration.

root@rob-laptop:~# /etc/init.d/apache2 restart

Set up MySQL
Before you can do anything to MySQL, you need to make sure it's running:

root@rob-laptop:~# /etc/init.d/mysql start

Starting MySQL database server: mysqld.

root@rob-laptop:~#

Now, you must cd to the /usr/src directory and get the source to freepbx using svn:

root@rob-laptop:/usr/src/freepbx# cd /usr/src/

root@rob-laptop:/usr/src# svn co https://svn.sourceforge.net/svnroot/amportal/freepbx/branches/2.2 freepbx-2.2

A freepbx/amp_conf

A freepbx/amp_conf/astetc

... freePBX now downloads ...

Checked out revision 2574.

root@rob-laptop:/usr/src# cd /usr/src/freepbx-2.2

Now, to configure the databases for freePBX:

root@rob-laptop:/usr/src/freepbx# mysqladmin create asterisk

root@rob-laptop:/usr/src/freepbx# mysqladmin create asteriskcdrdb

root@rob-laptop:/usr/src/freepbx# mysql asterisk < SQL/newinstall.sql

root@rob-laptop:/usr/src/freepbx# mysql asteriskcdrdb < SQL/cdr_mysql_table.sql

They also need to be secured, so that not just anyone can access
them. freePBX will prompt you for a database password when you do the
install. You need to pick that now. We'll assume that you've picked
'asteriskuser' and 'amp109' - you probably shouldn't use these, as they
are well known passwords for Asterisk@Home builds. If anyone's trying
to attack your machine, they will try this.

root@rob-laptop:/usr/src/freepbx# mysql

Welcome to the MySQL monitor. Commands end with ; or \g.

Your MySQL connection id is 12 to server version: 5.0.22-Debian_0ubuntu6.06.2-log

Type 'help;' or '\h' for help. Type '\c' to clear the buffer.

mysql> GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY 'amp109'; (This is the first username and password asked for below)

Query OK, 0 rows affected (0.00 sec)

mysql> GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY 'amp109';

Query OK, 0 rows affected (0.00 sec)

mysql> flush privileges;

Query OK, 0 rows affected (0.00 sec)

mysql> \q

Bye

root@rob-laptop:/usr/src/freepbx#

Now, after all of this, you need to pick a root 'mysql' password.
For this, we'll pretend it's 's33kret'. If you need to do anything else
with mysql, you'll need to provide this password.

root@rob-laptop:/usr/src/freepbx# mysqladmin -u root password 's33kret'

Install freePBX at last!
You're there - you've done the hard yards, and finally you can install freePBX!

WARNING! If you have an existing Asterisk
installation, the script below will overwrite your Asterisk
configuration files. Backup your /etc/asterisk directory before
running.

root@rob-laptop:/usr/src/freepbx# ./install_amp

Checking for PEAR DB..OK

Checking for PEAR Console::Getopt..OK

Checking user..OK

Checking for /etc/amportal.conf../etc/amportal.conf does not exist, copying default

Creating new /etc/amportal.conf

Enter your USERNAME to connect to the 'asterisk' database:

[asteriskuser] (Just push enter if you've done the defaults above, or, fill in the details you entered)

Enter your PASSWORD to connect to the 'asterisk' database:

[amp109] (As above, the password you picked in the MYSQL command)

Enter the hostname of the 'asterisk' database:

[localhost] (Just push enter)

Enter a USERNAME to connect to the Asterisk Manager interface:

[admin] (Just push enter)

Enter a PASSWORD to connect to the Asterisk Manager interface:

[amp111] (Just push enter)

Enter the path to use for your AMP web root:

[/var/www/html]

/var/www

Enter the path to use for your FOP web root:

[/var/www/html/panel]

/var/www/panel

Enter the path to your Apache cgi-bin:

[/var/www/cgi-bin] /usr/lib/cgi-bin

Enter the IP ADDRESS or hostname used to access the AMP web-admin:

[xx.xx.xx.xx] Enter the IP Address of your UBUNTU SERVER HERE

Enter a PASSWORD to perform call transfers with the Flash Operator Panel:

[passw0rd] (Just push enter)

Use simple Extensions [extensions] admin or separate Devices and Users [deviceanduser]?

[extensions](Just push enter)

Enter directory in which to store AMP executable scripts:

[/var/lib/asterisk/bin] (Just push enter)

Created /var/lib/asterisk/bin

Enter directory in which to store super-user scripts:

[/usr/sbin] (Just push enter)

/etc/amportal.conf writtenOK

Reading /etc/amportal.conf..OK

Checking for /etc/asterisk/asterisk.conf..OK

Reading /etc/asterisk/asterisk.conf..OK

Checking for Asterisk 1.2..OK

Checking for selinux..OK

At this stage, you're almost done, but there's quite often a
problem when people have made a typo, or forgotten to put a password in
the mysql server. If you see these lines:

Connecting to database..FAILED

Try running ./install_amp --username=user --password=pass (using your own user and pass)

[FATAL] Cannot connect to database

root@rob-laptop:/usr/src/freepbx#

it means that you haven't done the 'GRANT ALL PRIVILEGES ...'
command in MySQL, or, you've put the wrong password in when you were
doing the installation. All is not lost. If you want to re-run the
installation, with it prompting you again, you can simply delete the
/etc/amportal.conf file (rm /etc/amportal.conf) or edit it and put the correct password in. Then re-run ./install_amp and it should proceed along happily.

root@rob-laptop:/usr/src/freepbx# ./install_amp

Checking for PEAR DB..OK

Checking for PEAR Console::Getopt..OK

Checking user..OK

Checking for /etc/amportal.conf..OK

Reading /etc/amportal.conf..OK

Checking for /etc/asterisk/asterisk.conf..OK

Reading /etc/asterisk/asterisk.conf..OK

Checking for Asterisk 1.2..OK

Checking for selinux..OK

Connecting to database..OK

Checking current version of AMP..2.1.2

Installing new AMP files..OK

Configuring install for your environment..OK

Setting permissions on files..OK

Checking for upgrades..0 found

Generating AMP configs..

Generating Configurations.conf..

Checking for PEAR DB..OK

Checking for PEAR Console::Getopt..OK

Checking for /etc/amportal.conf..OK

Reading /etc/amportal.conf..OK

Reading /etc/asterisk/asterisk.conf..OK

Connecting to database..OK

Please Reload Asterisk by visiting http://192.168.1.53/admin

Generating AMP configs..OK

Restarting Flash Operator Panel..op_server.pl: no process killed

OK

Please Reload Asterisk by visiting http://192.168.1.53/admin

root@rob-laptop:/usr/src/freepbx# modprobe ztdummy

root@rob-laptop:/usr/src/freepbx# amportal start

Starting FreePBX and Asterisk automatically
If you don't have any zaptel hardware, you can automatically start
ztdummy and asterisk by editing /etc/rc.local, and before the 'exit 0'
line insert these two lines:

modprobe ztdummy
/usr/sbin/amportal start &

That ensures that the timing module (ztdummy) is loaded, and that
asterisk is running on bootup. You can also use the /etc/init.d/zaptel
script to start zaptel, hopefully someone with more ubuntu knowledge
can update this page with how to enable it.

Congratulations!
You're done - you now have a fully functional FreePBX
Installation. The first thing to do is log in (Go to the IP address of
the Ubuntu machine above, and click on 'Setup'). The default username
is 'admin' and the default password is 'admin'. Go to the
'Administrators' tab and change the password straight away. After that,
you can visit the Online Module Repository and see what modules are
available.

Install Process for Ubuntu Server 7.04

Placeholder until more detailed instructions can be written
For now, see UbuntuServer or the Ubuntu 6 Installation Instructions

Upgrading your system

Upgrading FreePBX
Note: You may need to install subversion for your linux distribution if you have any errors. Do this by typing yum -y install subversion at the root prompt.
Upgrading to the latest Released Version: 2.7.0
You will want to pull the release tarball, and follow the instructions below. Once you have installed it you will want to navigate to Module Admin menu item and update all modules as well as install other available modules not included in the release tarball.

cd /usr/src/
wget http://mirror.freepbx.org/freepbx-2.7.0.tar.gz
tar zxvf freepbx-2.7.0.tar.gz
cd freepbx-2.7.0
./start_asterisk start # for upgrades use: amportal start 
./install_amp

On some rare cases you may experience an issue with the SQL database not being correctly updated. If that appears to be the case, you can safely add the --force-version option. For example, if you were upgrading from version 2.1.3 you might type:

cd /usr/src/freepbx-2.7.0
./install_amp --force-version=2.1.3

(Replace 2.1.3 with the version you are upgrading from, or an earlier version which is also safe)

Upgrading to the latest stable 2.7 branch from SVN
If you prefer to pull directly from the SVN repository you can follow these instructions below. Once you have installed FreePBX it is important to navigate to Module Admin in FreePBX and upgrade your modules. Installing FreePBX in this way (vs. using the release tarball) is not recommended for initial installs unless you are very familiar with the project. Without loading critical modules like core, framework, voicemail and some others your system will not be able to do anything for you or will function improperly.

cd /usr/src/
svn co http://svn.freepbx.org/freepbx/branches/2.7 freepbx-2.7

Installation instructions are the same as above after exploding the tarball

Note:Need to get the new func_devstate features to take advantage of the optional BLF? You can get a version that works on 1.4 from this website, and then you simply put the file in your apps subdirectory of your asterisk source and re-run make and make install. To use this mode, amportal.conf needs the setting:

USEDEVSTATE=true

Upgrading to the current 2.8 Release Candidate
We just released 2.7 and 2.8 is being defined. It is currently trunk in SVN. We will update this section with more details as we role out the alpha/beta/RC program for 2.8 more formally in the coming weeks.
Upgrading to 2.6 From trixbox
If you are using a version of trixbox that no longer connects to the FreePBX repository and does not provide you with the 2.6 Upgrade Tool, you can upgrade to 2.6 by following these simple instructions. The process will simply install a special version of the Upgrade Tool and then you will upgrade just the same as the normal GUI upgrade process. Although the tool may indicate you are upgrading to a beta or release candidate program, since 2.6 has gone final, you will actually be upgraded to the current 2.6 released version. The steps are simply:

1. Download the upgrade module to your desktop by pressing this link: Download trixbox Upgrade Tool
2. Navigate to Module Admin in FreeBPX, click on the Upload module link and then browse to the module you downloaded in step 1, choose it, and then press upload.
3. Now you simply install the module as if you were downloading it from the Online Repository and then you can follow the instructions provided by the module to upgrade!

The process is almost identical to pulling the module from our repository and once you do this, you will have access to all the great features and many bug fixes no present elsewhere.
Upgrading to the trunk branch
The trunk is where active development happens. The project tries hard to keep it relatively stable and you will find it is fairly safe to run off of this branch most of the time, although it is actively being developed on so glitches can occur. This differs from release branches where they are limited to the latest bug fixes. You can access the development trunk as described below.
To run off of trunk and keep up-to-date with every change even before modules get published, there is a set of tools that will help you build an install environment where you do NOT use Module Admin to keep up-to-date. The instructions are:

cd /usr/src/
svn co http://svn.freepbx.org/freepbx/trunk freepbx-trunk
cd freepbx-trunk
./setup_svn.php

At this point, you will now have a special SVN environment where all the FreePBX modules are pulled into your install directory. You can then do a normal install:

./install_amp --force-version=2.4.1

Note that you will always want to do a force-version in case there are new schema changes introduced since the last time you updated. If you are updating from a release prior to 2.4.1, you should use that release number the first time, but 2.4.1 after that is fine.
Now, instead of using Module admin with this setup, you can simply follow the following procedure to update yourself to the newest updates that may have been committed:

cd /usr/src/freepbx-trunk
svn update
./install_amp --force-version=2.4.1 #(*)

(*) if there were no updates pulled, then there is no need to do an install amp again
The install will take longer because it will be going through the process of reinstalling all modules, but it will do such with all new changes even before they are published to the Online Repository.

Adding Extensions

Adding Extensions

A PBX without any extensions isn't very useful, so it's the first thing to do after installing FreePBX. Extensions let you test all kinds of things, so it's the first thing to get right.

Shown at right are a few test extensions on a FreePBX installation on my t42 Ubuntu laptop.

There are several pages of information here. We'll go through each of them.

Display Name: This is the name that is used, at least internally, when placing an outbound call. Most Caller Name services look up the name in a database, so this name setting might do nothing on your outbound VOIP or PRI calls. It will certainly do nothing on outbound POTS calls.

CID Num Alias: The CallerID to show when dialing intracompany. Example Usage: James has a office extension at 201, a softphone at 401, a home office phone at 601, and a FollowMe at 201 that rings them all. 401 and 601 can use a CID Num Alias of 201, so that all internal call recipients see “201”

SIP Alias: Every 'clever' presentation of VOIP has an example of dialing by email address. This is hard to do on most phones, but is nonetheless supported. Put only the name here, not the @ symbol or the fully-qualified-domain name. That's used by the calling application or device to locate your PBX on the internet. To allow any party to call you, you'll need to have firewall rules that allow all SIP calls regardless of IP address. This is only advisable if your Asterisk installation is up-to-date, and has no current SIP security vulnerability.

Direct DID: This is where you enter the Direct Inward Dial (DID) you'd like to reach this extension. If you forget, all calls to that DID will end up at the main IVR. Putting a value here eliminates the need to create an Inbound Route.

DID Alert Info: Used for distinctive ring services

Music on Hold: Set a different Music On Hold (MOH) class for this extension. Great for having different music for different offices or companies that are served by the same PBX.

Outbound CID: Put the CallerID and preferred CallerIDName here for outbound usage.

Ring Time: How long to ring before a server-side transfer to voicemail. You'll usually use the default here, and set a system-wide value in General Settings.

Call Waiting: Set the call waiting value. Also accessible by feature code from an individual extension (by default *70 to activate and *71 to deactivate – see Feature Codes).

Emergency CID: The CallerID to be set when dialing a number labeled as emergency.

Device Options

Extensions - Device Options

These options are the same as in a vanilla asterisk sip.conf file. In a FreePBX installation, they end up in sip_additional.conf. For more information, check out Asterisk: TFOT.

secret: The SIP password used in the authentication of this device to the server.

dtmfmode: How DTMF is expected by the server. Options are rfc2833, INFO, and in-band. rfc2833 seems the most reliable across many devices. Client devices (e.g. Linksys) often have an Auto setting, which is to be avoided.

canreinvite: Asterisk is a back-to-back useragent. This means that your phone calls it, and it calls your VOIP, PRI or POTs line. All audio (RTP stream) is carried through the Asterisk process during the call. Your VOIP service provider, for example, often will use a SIP REINVITE message to change the RTP destinations after the call is set up. This reduces load on the equipment, as it's only doing call setup and takedown.

Highly desireable if you're supporting remote users making VOIP calls and your VOIP provider supports REINVITE.

However, it's tricky to get any of your FreePBX features to work in this scenario. Play with this, but don't use it on a customer system unless you have tested the features you need.

context: Context is an Asterisk dialplan sphere-of-influence concept used to separate components from each other (multi-tenant, for example, or outward facing customer service from backoffice).

From-internal means you can dial like you're a phone on premesis with access to other extensions and outbound trunks. Other common options are outbound-all-routes (dial out only), from-trunk (extensions only, no outbound dialing)

host: dynamic or a static IP address. dynamic allows any device that can pass the SIP challenge/authentication to register and make/receive calls.

type: friend or peer. Use friend for a phone. Peer is for SIP devices that are capable of carrying calls, like a Trunk.

nat: yes or never. SIP is a nat-unfriendly protocol in that it specifies the return IP address for the call audio stream deep inside a packet. NAT works by rewriting packet source and destination IP addresses, but doesn't understand SIP (unless a good SIP Application Layer Gateway is installed). NAT is therefore problem if both the phone and the server PBX are separated from the public internet by different NATs (e.g. a home router and and corporate one.) In such a situation, audio won't work, but signaling will (phones will ring but no audio). To support remote home users behind conventional NATs, use yes, and either give the server PBX a public IP address or do a 1:1 IP mapping from a public IP to it's internal, then set IP_nat.conf to the public IP address of the system. NAT=yes instructs Asterisk to send audio to the IP it receives it from, regardless of what the SIP SDP says, and lets you have at least one NAT present and still have effective audio. Note that NATs vary widely as to how long they stay 'open'. Best practice when using Non-STUN phones is to have SIP registration expire every 60 seconds – the re-registration (outbound, by the phone) will keep the NAT open to receive calls. NAT=yes doesn't hurt anything when the client device is on the same LAN.

callgroup:

pickupgroup:

disallow: enter codec overrides here. An extension or group of extensions on a low-bandwidth link might want to disallow the higher-bandwidth codecs out of the general pool.

allow: enter any codc overrides here

dial: SIP/extension is the default.

accountcode: enter an account code for use by a billing module.

mailbox: extension@default is the default.

Asterisk CLI Commands

General commands
!<command>: Executes a given shell command 

abort halt: Cancel a running halt

add extension: Add new extension into context

add ignorepat: Add new ignore pattern

add indication: Add the given indication to the country

amportal start: Stop AAH and

amportal stop: Restart AAH.

debug channel: Enable debugging on a channel

dont include: Remove a specified include from context

help: Display help list, or specific help on a command

include context: Include context in other context

load: Load a dynamic module by name

logger reload: Reopen log files. Use after rotating the log files.

no debug channel: Disable debugging on a channel

pri debug span: Enables PRI debugging on a span

pri intense debug span: Enables REALLY INTENSE PRI debugging

pri no debug span: Disables PRI debugging on a span

remove extension: Remove a specified extension

remove ignorepat: Remove ignore pattern from context

remove indication: Remove the given indication from the country

save dialplan: Overwrites your current
extensions.conf file with an exported version based on the current
state of the dialplan. A backup copy of your old extensions.conf is not
saved. The initial values of global variables defined in the [globals]
category retain their previous initial values; the current values of
global variables are not written into the new extensions.conf. Using
"save dialplan" will result in losing any comments in your current
extensions.conf.
set verbose: Set level of verboseness

show agents: Show status of agents

show applications: Shows registered applications

show application: Describe a specific application

show channel: Display information on a specific channel

show channels: Display information on channels

show codecs: Display information on codecs

show conferences: Show status of conferences

show dialplan: Show dialplan

show image formats: Displays image formats

show indications: - Show a list of all country/indications

show locals: Show status of local channels

show manager command: Show manager commands

show manager connect: Show connected manager users

show parkedcalls: Lists parked calls

show queues: Show status of queues

show switches: Show alternative switches

show translation: Display translation matrix

show voicemail users: List defined voicemail boxes

show voicemail zones: List zone message formats

soft hangup: Request a hangup on a given channel

A.2.2 AGI Commands
show agi: Show AGI commands or specific help

dump agihtml: Dumps a list of agi command in html format

A.2.3 Database Handling
database del: Removes database key/value

database deltree: Removes database keytree/values

database get: Gets database value

database put: Adds/updates database value

database show: Shows database contents

A.2.4 IAX Channel Commands
iax2 debug: Enable IAX debugging

iax2 no debug: Disable IAX debugging

iax2 set jitter: Sets IAX jitter buffer

iax2 show cache: Display IAX cached dialplan

iax2 show channels: Show active IAX channels

iax2 show peers: Show defined IAX peers

iax2 show registry: Show IAX registration status

iax2 show stats: Display IAX statistics

iax2 show users: Show defined IAX users

iax2 trunk debug: Request IAX trunk debug

iax debug: Enable IAX debugging

iax no debug: Disable IAX debugging

iax set jitter: Sets IAX jitter buffer

iax show cache: Display IAX cached dialplan

iax show channels: Show active IAX channels

iax show peers: Show defined IAX peers

iax show registry: Show IAX registration status

iax show stats: Display IAX statistics

iax show users: Show defined IAX users

init keys: Initialize RSA key passcodes

show keys: Displays RSA key information

A.2.5 SIP Channel commands
sip debug: Enable SIP debugging

sip no debug: Disable SIP debugging

sip reload: Reload sip.conf (added after 0.7.1 on 2004-01-23)

sip show channels: Show active SIP channels

sip show channel: Show detailed SIP channel info

sip show inuse: List all inuse/limit

sip show peers: Show defined SIP peers (register clients)

sip show registry: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy)

sip show users: Show defined SIP users

A.2.6 Server management
restart gracefully: Restart Asterisk gracefully

restart now: Restart Asterisk immediately

restart when convenient: Restart Asterisk at empty call volume

reload: Reload configuration

stop gracefully: Gracefully shut down Asterisk

stop now: Shut down Asterisk immediately

stop when convenient: Shut down Asterisk at empty call volume

extensions reload?: Reload extensions ONLY

unload: Unload a dynamic module by name

show modules: List modules and info about them

show uptime: Show uptime information

show version: Display Asterisk version info

Connecting 2 or more boxes

There may be a time when you want to interconnect 2 Asterisks boxes
(def.com.au and xyz.com.au) together and if you are like me, you will
probably be spending a good part of 3 hours trying to get them to talk
to one another.

I have 2 different locations, the Main Office (def.com.au) with
about 11 extensions and another office in a different location
(xyz.com.au) about 20 km away with 9 extensions. The main office is the
only box that will have accounts with different VSPs and all external
communications are through the main office Asterisk box. I settled for
the simplest solution and after some fiddling around I managed to get
them to work the way I wanted it but not happy with it, I solicited
some advise from a friend (thanks to Mark Brooker) who told me that my
configuration could be made a lot tidier. That I did.

Instead of being verbose in my explanation, I will just create a
few tables outlining what I did. I hope this will help those in the
same position as I am, to set 2 very basic systems together (you can
refer to DUNDi for a more complete solution).

27.1 METHOD 1 - with the peer Asterisk boxes as extensions
For the purpose of registering the peers to each other, I created
1 extension on each box eg: 90000 on System 1 and 91000 on System 2–
using extension numbers that I am not likely to use as local extensions
(while some users have had success using common extension, but I prefer
2 separate extensions as I have them working). For simplicity, I gave a
common password xxxyyy to both boxes. Avoid using extension starting
with 8 as it may clash with conferencing.

 
System 1
System 2

IAX Trunk
 
 

Outgoing Dial Rules:
XX.
XX.

Trunk Name
Parramatta
MainOffice

Peer Details
host=xyz.com.au (or IP)

secret=xxxyyy

type=peer

username=91000

host=def.com.au (or IP)

secret=xxxyyy

type=peer

username=90000

User Context
Leave blank
Leave blank

User Details
Leave blank
Leave blank

Register String
80000:xxxyyy@xyz.com.au
90000:xxxyyy@def.com.au

Note: Registration isn’t really necessary. It will still work without it unless you use Dynamic IP.

 
System 1
System 2

Extensions
 
 

Phone Protocol
IAX
IAX

Extension Number
90000
91000

Extension Password
xxxyyy
xxxyyy

Fullname
Parramatta
Main Office

Voicemail & Directory
Disabled
Disabled

 
System 1
System 2

Outbound Routing
 
 

Route Name
Parramatta
MainOffice

Route Password
Leave Blank
Leave Blank

Dial Patterns
6XXX(6001 to 6009 are Parramatta Office extensions)
XX.(Apart from Local extensions, all others go via City Office)

Trunk Sequence
IAX2/Parramatta
IAX2/MainOffice

The above Outbound Routing rule assumes that you do not wish to use
a dialling prefix. If you want to use a prefix to dial the remote
extensions and to use the remote routing rules, you may place a prefix
e.g. 9|6XXX and 9|XX. for system 1 and system 2 respectively instead of just 6XXX and XX.

The above example assumes that both Asterisk boxes have Public Fix
IP address. If you have Dynamic IP addresses, you will need to register
both the boxes with DynDns to obtain a valid DNS ID. If you are a part
of a Corporate LAN, than you will have no need to worry about DynDns
and what not.

Note: While this method will provide some
rudimentary security (though pretty weak), as it requires an extension
to be created for the peer Asterisk box, it will not pass the calling
party extension number to the remote Asterisk box. Instead, it will
pass the Trunk ID only and all calls will seem to come from the same
trunk and not individual extension – I did say that this is a simple
solution.

27.2 METHOD 2 - In a Peer/User arrangement
Another method that I use is described below. This method treats
both the Asterisk box as internal to each other as peer and user. I am
using IAX2 for this purpose, however I believe, you may be able to do
this with SIP as well if you are trying to connect the older Asterisk
with the newer incarnations (I have not proved it yet). This method
does not require registration either and does not require you to create
extensions for the peers. In many ways, this is simpler to set up.

Unlike the first method, this second method will pass the Caller ID
to the receiving party. The receiving party will actually get the
callers’ extension number/ID instead of the extension number of the
peer Asterisk box.

Note: You must provide for security, as this is pretty wide open.

Like all installation, you must provide for security. As different
installation resorts to different types of security arrangement, I will
leave that to the individual implementer to deal with the security
issues.

(Note: A little tutorial on DUNDi can be found here).

Rather than being verbose, I will illustrate this method using tables as follows;

 
System 1
System 2

IAX2 Trunk
 
 

Outgoing Dial Rules:
6XXX
XX.

Trunk Name
InterOffice
InterOffice

Peer Details
host=xyz.com.au (or IP)

Qualify=yes

type=peer

host=def.com.au (or IP)

Qualify=yes

type=peer

User Context
InterOffice-In
InterOffice-In

User Details
context=from-internal

host=xyz.com.au (or IP)

type=user

context=from-internal

host=def.com.au (or IP)

type=user

 
System 1
System 2

Outbound Routing
 
 

Route Name
InterOffice
InterOffice

Route Password
Leave Blank
Leave Blank

Dial Patterns
6XXX(6001 to 6009 are Parramatta Office extensions)
XX.(Apart from Local extensions, all others go via City Office)

Trunk Sequence
IAX2/InterOffice
IAX2/InterOffice

Thinking of more than 2 boxes?
Just as a matter of interest, you can connect several boxes using
this method. While I have connected 3 boxes successfully,I believe, se
same principle can be applied to more boxes.

In my implementation I have box A, B and C (System 1, 2 and 3). Box
A is the master box. All the other boxes use box A as the main
exchange.

A peers with B and C - B peers with A - And C peers with A.

Except for local traffic, all external and inter-office
(inter-branch) traffic goes via Box A. – with the appropriate dial plan
of course.

Both the above methods, while useable for a basic configuration,
will not provide you with a complete solution. To provide a complete
solution is beyond the scope of this document.

The following link will provide further reference for connecting two Asterisk boxes together http://www.voip-info.org/wiki/view/Asterisk+dual+servers

If you require a complete solution tailored to your exact requirement, my advise to you is to hire a VOIP consultant.

Creating Administrator Roles

Creating Administrator Roles

For most web applications it is useful to have graduated
permission access, so that users have only access to the functions
they need.

This lets you give office managers, for example, access to the
Extensions directory to change usernames and reset voicemail
passwords as employees come and go, without exposing trunks and other
settings they do not need.

Show below is just such a configuration. In this case, Mie is
allowed to see status, edit extensions (this part is not shown) and
apply changes.

In addition to the webapp username / password settings, both
Apache and iptables can be used to restrict access on a location
basis to the web application.

A good policy is to only allow local (LAN) or tunneled via SSH
access to the web application, though exceptions can be be made for
the Recordings (ARI) interface.

Creating an IVR

Digital Receptionist or IVR

Information

The 'Digital Receptionist' page is the interface used to setup your auto attendant when people call your PBX. Normally heard as "Thanks you for calling MYBUSINESS, for Sales press 1, for Service press 2", etc.

Planning

While the urge is strong just to dive in by clicking on IVR, you should resist this impulse.

First, draw out on paper what you intend to to achieve. Run it by the customer (or your officemates). Write out word-for-word what all the recordings are going to be.

The proper flow to build a good IVR is:

1. Planning

2. Customer agreement with the plan.

3. Record the audio prompts using System Recordings and an extension.

4. Create any destinations that don't currently exist (queues, ring groups, day/night modes or time conditions).

5. Test all of these. One way to do this is use miscellaneous destinations, assigning a * feature code to whatever thing you want to test.

6. Then go create your IVR.

7. Show it to the customer, and then make the inevitable changes.

8. Now upgrade the voice prompts to a paid voice or designated employee (the office manager or receptionist, etc.)

9. Bask in glory!

Standard IVR Examples:

1. Office / Light industrial

   1. Welcome to BUSINESSNAME. Please listen carefully as our options have changed. If you know the extension of the person you are trying to reach, you may dial it at any time. Press 1 for sales, press 2 for customer service, press 3 for administration, press 4 for Press inquiries, press 5 for office directions,press # to access the company directory, or press 0 for the operator.

2. Hospitality

   1. Welcome to HOTELNAME. Please listen carefully as our options have changed. If you know the room # of the guest you are trying to reach, you may dial it at any time. Press 1 for reservations, press 2 for the front desk, press 3 for event sales, press 4 for hotel administration, press 5 for hotel directions, press # to access the hotel directory, or press 0 for the operator.

3. Engineering/Product Company with Direct Sales and Support

   1. Welcome to BUSINESSNAME. Please listen carefully as our options have changed. If you know the extension of the person you are trying to reach, you may dial it at any time. Press 1 for sales, press 2 for customer service, press 3 for technical support, press 4 for administration, press 5 for Press inquiries, press 6 for office directions, press # to access the company directory, or press 0 for the operator.

4. Retail

   1. Welcome to BUSINESSNAME. Please listen carefully as our options have changed. If you know the extension of the person you are trying to reach, you may dial it at any time. Press 1 for sales, press 2 for customer service, press 3 for store hours, locations, and directions, press 4 for administration, press 5 for Press inquiries, press # to access the company directory, or press 0 for the operator.

Making recordings

Fire up the System Recordings module. Shown here is 3.3.5.1.

I strongly suggest you use an extension connected to the PBX to make your recordings. They'll be quick and in the right format and you can worry about getting everything else right. When everything is all finished, you can come back and replace those temporary recordings with paid or improved versions.

To use your extension to make a recording, enter your extension in Step 1 and press Go. Don't skip this and go to Step 2, or you'll get a cryptic error.

Now dial *77 and make your recording after the beep. Dial *99 to listen to it. You don't have to be the person doing this – I often enter a customer's extension and have a customer do this part while I do the GUI work.

If the recording is good enough (and don't obsess here yet), name the recording and press Save.

For lame and silly reasons, spaces are not allowed in the names.

You can listen to your recording and add on other recordings (such as the built-in recordings) by clicking on your recording in the right tool panel.

We're going to start with a simple 1-level IVR , so the single Welcome-to-ACME recording will be enough.

Now that we've created a system recording, we can create our IVR.

Creating the IVR

When you select IVR, the first page is now a brief set of instructions on how to drive the IVR. You can either edit an IVR, if one is existing, or create a new one by clicking on 'Add IVR'.

Editing your IVR

Unlike the old Digital Receptionist system, this creates the IVR (and calls it 'Unnamed') as soon as you click 'Add' - You'll see it appear on the right straight away.

These are your options:

• Change Name: This is simply the descriptive name that appears on the right, and in the drop-down menu of Destinations

• Timeout: This is the amount of time the system waits before sending the call to the 't' destination

• Enable Directory: If you switch this on, users will be able to dial the FeatureCodes">feature code for Directory, usually #, from the IVR and access the Directory service.

• Directory Context: This is the asterisk context of the directory. Advanced users can then use different IVRs to create a multi-tenant installation.

• Enable Direct Dial: If you enable that, users will, in addition to being able to dial the IVR options, be able to directly dial an Extension number.

• Announcement: A System Recording that is played to users when they enter the IVR. This can be set to 'nothing'. These announcements are great for “today is July 4^th and we're closed for the holiday” and then proceeding on to the regular call flow.

Configuring your IVR

In the box on the left, enter the option for the user. This may be one, or a series of numbers, or, 'i', or 't'. 'i' and 't' have special meanings:

• i: This overrides the default invalid choice behavior, which is to play a 'invalid option' message and immediately replay the current menu. E.G. If you only have 1 2 and 3 defined, and caller pushes 4, it will jump to this destination.

• t: This overrides the default timeout behavior, which is to play the menu three times and hangup. A standard configuration is to go the operator, to handle customers that don't have DTMF-capable phones.

Options are only displayed if there is at least one entry created. For example, queues will not appear as a possible IVR destination if no queues exist.

Use 'Increase Options' or 'Decrease Options' to alter the number of options available. This won't let you decrease it to less than the number of options that are currently set.

To delete an option, simply leave the selection blank.

When you're finished, click 'Save' and you have your new IVR.

To test it, give it an incoming route or set up a miscellaneous application (* code) to reach it.

Creating and Assigning Extensions

Creating and Assigning Extensions

Numbering Schemes

There are several schemes for assigning extensions. Invariably, though, you'll find the following guidlines will help:

• Use their previous extension numbers
  • Upgrading a system shouldn't require upgrading business cards
• Use the last 3 or 4 digits of their DIDs
  • Less for people to remember
• For non-DID systems, choose the last 3 digits of the main number
  • If the main number is 651-3200, then extensions can be 200, 201, 202, etc.
• Don't collide with system shortcuts, common dialing sequences, or emergency numbers
  • In the US, this rules out extensions in the 100s and 900s, at minimum. 611 and 311 shouldn't be assigned, but the rest of the 600s and 300s can be.
  • For FreePBX, 7777 is commonly 'simulate an incoming call', should be avoided, as should other Miscellaneous Destinations

Remember, when reserving DIDs, to get the whole block of interest if possible. It's usually low enough cost, and it really hurts to run out.

File ownership and what files you can edit

Who owns what files in /etc/asterisk when FreePBX is installed?
That's what this page is here to answer.
The basic rule is that all files are owned and modified by FreePBX unless they end _custom.conf. There are a few exceptions to this rule but not many.
If the file is owned by FreePBX you should find this statement at the top of the file making it clear that it is owned by FreePBX

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

So here is the list of files as of version 2.4. Those owned by FreePBX will be in bold underline. If they become owned in a later version that version will be stated to the right of the file name.
agents.conf
alarmreceiver.conf
applications.conf
asterisk.conf
backup.conf

1. This file contains the crontab line(s) that will get executed for backup job scheduling.

cdr_mysql.conf

1. if you want to use the userfield in the CDR reporting you will need to add this line to the file: userfield=1
   then restart Freepbx by typing amportal restart
   Default file should look like this:

    ;
    ; Note - if the database server is hosted on the same machine as the
    ; asterisk server, you can achieve a local Unix socket connection by
    ; setting hostname=localhost
    ;
    ; port and sock are both optional parameters.  If hostname is specified
    ; and is not "localhost", then cdr_mysql will attempt to connect to the
    ; port specified or use the default port.  If hostname is not specified
    ; or if hostname is "localhost", then cdr_mysql will attempt to connect
    ; to the socket file specified by sock or otherwise use the default socket
    ; file.
    ;
    [global]
    hostname=localhost
    dbname=asteriskcdrdb
    password=amp109
    user=asteriskuser
    ;port=3306
    ;sock=/tmp/mysql.sock
   

codecs.conf
dnsmgr.conf
dundi.conf
enum.conf
extconfig.conf
extensions.conf

1. if you need to modify existing code code/context in extensions.conf please place your modifications in extensions_override_freepbx.conf as asterisk uses the code for the first context referance and ignores additional occurances.

extensions_additional.conf

1. DO NOT EDIT THIS FILE, it get's regenerated each and every time you apply changes.
2. If you need to expand on functionality of a section of code check to see if there is a include context line in the code (will end in _custom.conf) if so create that context in extensions_custom.conf and it will get called.
3. If you need to replace the functionality in extensions_additional.conf please place it in extensions_override_freepbx.conf but read the notes about this file first.

extensions_custom.conf

1. this is the file that you place all your custom contexts, and additional code enhancements to the FreePBX dial plan. This file will not be overwritten.

extensions_override_freepbx.conf

1. If extensions.conf (or extensions_additional.conf) has a context or macro that you NEED to modify, you place that code here as asterisk will only execute the first occurrences of that code and ignores other occurrences. This file will not be overwritten. Be very careful as replacing an existing piece of code this way is the fastest possible way to break your system. If you are doing this you should probably think about filing for a feature request or bug fix to get it addressed properly.

features.conf
features_applicationmap_additional.conf
features_applicationmap_custom.conf
features_featuremap_additional.conf
features_featuremap_custom.conf
features_general_additional.conf
features_general_custom.conf
globals_custom.conf
iax.conf
iax_additional.conf
iax_custom.conf
iax_custom_post.conf
iax_general_additional.conf
iax_general_custom.conf
iaxprov.conf
iax_registrations.conf
iax_registrations_custom.conf
indications.conf
localprefixes.conf
logger.conf
manager_additional.conf
manager.conf
manager_custom.conf
meetme.conf
meetme_additional.conf
mgcp.conf
modem.conf
modules.conf
musiconhold_additional.conf
musiconhold.conf
musiconhold_custom.conf
oss.conf
parking_additional.inc (should no longer be used as parking was moved to features)
phone.conf
phpagi.conf
privacy.conf
queues.conf

1. Do not edit this file in any way. Anything you can think of putting in this file can be placed into one of the _custom.comf files where it will not get removed or replaced.

queues_additional.conf

1. Do not edit this file in any way. Anything you can think of putting in this file can be placed into one of the _custom.conf files where it will not get removed or replaced.

queues_custom.conf

1. This is the proper location for placing any of the context specific options and lines that you might need to add before the processing of the queues_additional.conf file for your queues setup.

queues_custom_general.conf

1. This is the proper location for placing any of the [general] context option lines that you might need to add to your queues setup.

queues_general_additional.conf

1. Do not edit this file in any way. Anything you can think of putting in this file can be placed into one of the _custom.comf files where it will not get removed or replaced.

queues_post_custom.conf

1. This is the proper location for placing any of the context specific options that you might need to add to the end queues setup.
   This is the file that allows you to add or remove values to those entries found in the auto-generated queue_additional.conf file. So for example you have a queue 79 that need a additional parameter added. create a context line: [79](+) then on the next line add the item(s) you need to add. To remove use (-) instead followed by the line(s) you want removed.

res_mysql.conf
rtp.conf
sip.conf

1. Do not edit this file in any way. Anything you can think of putting in this file can be placed into one of the _custom.comf files where it will not get removed or replaced. If you are looking to do nat'ing, see sip_general_custom.conf or if it is a legacy system sip_nat.conf. If you want to add additional setup parameters for your sip device see sip_custom_post.conf, etc. If you need to adjust sip jitter or something else it will be sip_general_custom.conf (if it is for the general context) or sip_custom.conf. If you do edit this file and place something new in it, it will get overwritten at some point and next time you restart your system you will suddenly wonder why things stopped working.

sip_general_additional.conf

1. This is where FreePBX places all of it's general context settings. If you need to override one of these or add a new one please do so in sip_general_custom.conf.

sip_general_custom.conf

1. This is the proper location for placing any of the [general] context option lines that you might need to add to your setup. This is also the place to add those lines needed to enable the nat'ing of SIP when you go through a firewall.

   Some of the required lines for nat'ing are externip=, nat=, localnet= (you can have more then one occurrence of this line), and optionally fromdomain=. The first three are needed to properly setup a box on protected network behind a firewall that is providing nat to a public IP. If you have a legacy system these lines might have been placed in sip_nat.conf in the past, if so that is ok as long as the lines only exist in one file and not both (or a big debugging mess will occur along with hair loss as you pull it out while tracking it all down). See sip_nat.conf for more info.

   configurations with multiple subnets:
   For those setups with internal networks that have multiple subnets you will need to add a localnet= line for each subnet that the phone system should have direct access to. If you don't do this the phone system will assume that phones on those other subnets are external and thus provide the External IP of the box in the SIP headers instead of the internal IP. This then becomes a routing problem for the phone as it should not be attempting to talk external IP of the internal box (most firewalls can not handle the looping back of IP traffic).

   Example:
   Server 192.168.1.2 on a 192.168.1.0/255.255.255.0 network
   Phones inside the office are on the 192.168.2.0/255.255.255.0 subnet

   Requires these two lines in the either sip_general_custom.conf or sip_nat.conf file
   localnet=192.168.1.0/255.255.255.0
   localnet=192.168.2.0/255.255.255.0

sip_nat.conf

1. This is the old common location for placing the lines needed to enable the nat'ing of SIP. The new preferred location is sip_general_custom.conf. If you move the lines from this file to sip_general_custom.conf please remove them from this file or you'll experience hair loss as you spend time debugging why things don't work as you expect.

sip_registrations.conf

1. General section registrations that are auto-generated by FreePBX.

sip_registrations_custom.conf

1. a custom file just in case there is ever a need to override a general registration that was auto-generated by FreePBX.

sip_custom.conf

sip_additional.conf

1. This is where FreePBX puts all sip extensions, sip trunks, etc. If you need to add a additional parameter to a extension, trunk, etc., see sip_custom_post.conf.

sip_custom_post.conf

1. This is the file that allows you to add/remove values to those entries found in the auto-generated sip_additional.conf file. So for example you have an extension 1000 that needs an additional parameter added. Create a context line: [1000](+) then on the next line add the item(s) you need to add. To remove use (-) instead followed by the line(s) you want removed.

sip_notify.conf
skinny.conf
voicemail.conf

1. This file is both editable by you and by FreePBX, so please be careful. The structure of this file is as follows:

   [general]
   #include vm_general.inc
   #include vm_email.inc
   [default]
   

   Once you have configured a system with voicemail there will be values after the context [default]. These lines will be generated by FreePBX every time you add/edit/delete a extension.

   If you are looking to customize the e-mail message that get's send out with a voice mail please edit the vm_email.inc file. If you need to edit the mail sending parameters edit the vm_general.inc file. 99% of the world needs to edit two lines in the vm_general.inc file at the initial build time.

   The most common change to this file is to create a context called [zonemessages]. This context allows you to create timezones so that when you have extensions in multiple time zones they can date time stamp recorded messages properly for any given extension. If you create this context it should be placed after the second #include line and before the [default] line.

   [general]
   #include vm_general.inc
   #include vm_email.inc
   [zonemessages]
   eastern =       America/New_York|'vm-received' q 'digits/at' IMp
   central =       America/Chicago|'vm-received' q 'digits/at' IMp
   mountain =      America/Denver|'vm-received' q 'digits/at' IMp
   pacific =       America/Tijuana|'vm-received' q 'digits/at' IMp
   eastern24 =     America/New_York|'vm-received' q 'digits/at' R
   central24 =     America/Chicago|'vm-received' q 'digits/at' R
   mountain24 =    America/Denver|'vm-received' q 'digits/at' R
   pacific24 =     America/Tijuana|'vm-received' q 'digits/at' R
   deutschland =   Europe/Berlin | 'vm-received' Q 'digits/at' kM
   england =       Europe/London | 'vm-received' Q 'digits/at' R
   germany =       Europe/Berlin | 'vm-received' Q 'digits/at' kM
   alberta =       Canada/Mountain | 'vm-received' Q 'digits/at' HM
   madrid =        Europe/Paris|'vm-received' Q 'digits/at' R
   paris   =       Europe/Paris|'vm-received' Q 'digits/at' R
   sthlm   =       Europe/Stockholm|'vm-recieved' Q 'digits/at' R
   europa  =       Europe/Berlin|'vm-received' Q 'digits/at' kM
   italia  =       Europe/Rome|'vm-received' Q 'digit/at' HMP
   military = Zulu | 'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
   
   [default]
   
   

vm_email.inc

1. this file contains the e-mail subject line and message body for any voice mails that are e-mailed.

vm_general.inc

1. this file contains the e-mail / voice mail configuration parameters.
2. The most common change to this file is to edit the servermail= line so that it is from a valid worldly e-mail address or any mail server that has spam and/or spoofing protection will reject the voice mail e-mails.
3. other common lines to edit are: maxmessage= this is the max message limit, maxmsg= limits the total number of messages allowed in a mailbox, operator= if this is set to yes then when a person is leaving a message they can press 0 for the operator (or dial another extension).

zapata.conf
zapata-auto.conf
zapata_additional.conf
zapata_custom_chan_default.conf

Hardware examples

Add child pages to enter hardware examples here.
Make sure and note what call levels (and conferences, etc.) the system acheives.

High Quality Sounds

For those using the sounds that come with asterisk, you'll know that
volume and timing can be a bit wonky sometimes, and the quality isn't
all that great, due to them only being in GSM format. Kristian
Kielhofner of astLinux has come to the rescue by paying, out of his own pocket, to have all of the asterisk sounds re-recorded, and has released them under the BSD-License for all to use.

Here are the links to the files:

• aLaw Sounds (For use in most Countries)
• uLaw Sounds (For use in the US)
• GSM Sounds
• iLBC Sounds
• g729 Sounds
• S-Linear Sounds (The Asterisk Native format)
• ALL FILES above in one archive for easy installation - 26MB Download.

Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:

• Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer.
• Redistributions in binary form must reproduce the above
  copyright notice, this list of conditions and the following disclaimer
  in the documentation and/or other materials provided with the
  distribution.
• Neither the name of the University of California, Berkeley
  nor the names of its contributors may be used to endorse or promote
  products derived from this software without specific prior written
  permission.

THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS''
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS AND CONTRIBUTORS
BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

Interfacing to a PSTN

9.1 DIGIUM WILDCARD X100P FXO PCI CARD
This card allows you to connect a POTS (plain Old Telephone
System) line to your Asterisk@Home box (See Notes for Patch
information).

If this card is added after Asterisk has been configured, it may be
necessary to configure it by using the zaptel card auto-config utility
so the correct zaptel driver will be set up. To do that, enter the
following from the command line.

rebuild_zaptel (restart after each command)

genzaptelconf (see notes re command switch)

Next go into the AMP web interface to create a trunk and you will
notice that there is already a trunk called ZAP/g0. You need to edit
this.

1. Enter the phone number for you pots line in the Caller ID field
2. Enter 1 for Maximum channels
3. Set a dial rule you want for this trunk
4. Select an outbound dial prefix to select this trunk when dialing
5. Set the Zap Identifier to 1 (the default is g0)

Once the card is configured, you must add a route for Incoming Calls or asterisk will not answer this line

Click on Incoming Calls in AMP and set up an incoming route. To
make outbound calls you will need to set an outbound route as well.

If you have this card installed, you will need to edit the
following files; zapata.conf, zaptel.conf and modules.conf for AAH 1.x
or modprobe.conf for AAH 2.x. The last 2 files live in the /etc
directory – use a text editor to edit them.

9.1.1 zapata.conf
Under [channels] edit the following lines:

[channels]

busydetect=yes

busycount=6

For my installation to function correctly, I have also changed the
following setting to obtain a good compromise on volume/echoing:

rxgain=10.0 (you may have to experiment a little with this setting)

txgain=8.0 (you may have to experiment a little with this setting)

Ensure the following exist in zapata.conf. It is located at the end of the file.

;Include AMP Configs

channel => 1

#include zapata_additional.conf

Leave the rest of the file as it is.

9.1.2 zaptel.conf
Change the loadzone and defaultzone to 'au'

# Global data

loadzone = au

defaultzone = au

9.1.3 modules.conf (modprobe.conf for AAH 2.x)
For AAH 1.x, locate the post-install wcfxo entry and edit it to reflect this:

post-install wcfxo /sbin/ztcfg opermode=AUSTRALIA

For AAH 2.x, add the line highlighted in Bold below:

.

alias char-major-196 torisa

options wcfxo opermode=AUSTRALIA ; add this line

install tor2 /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg

.

9.2 DIGIUM TDM400P FXO/FXS CARD
Like the Digium Wildcard X100P, this card allows you to connect a
POTS (plain Old Telephone System) line to your Asterisk@Home box.
Unlike the X100P, this card has 4 module ports that can be loaded with
FXS or FXO modules. Channel 1 is the top RJ-45 on the back of the
TDM400P card.

If this card is installed after Asterisk has been loaded, you will
need to configure it just like the X100P by using the following command
on the command line:

genzaptelconf

9.2.1 zapata-auto.conf
Next, using config edit, look in the zapata-auto.conf file and you
will see a list of all your channels in your TDM400P. Set up the trunks
as trunks and the extensions as extensions in AMP.

When you open the zapata_auto.conf file, it will look something like the illustration below (see the red highlight)

zapata-auto.conf

; Span 1: WCTDM/0 'Wildcard TDM400P REV E/F Board 1'

signaling=fxo_ks

; Note - this is an extension. Create a ZAP extension in AMP for Channel 1

channel => 1 < - -this would have been defined already by the config

signaling=fxs_ks

;Note - this is a trunk. Create a ZAP trunk in AMP for Channel 2

context=from-pstnchannel => 2 < - -this would have been defined already by the config

If in the illustration it shows channel 1 is your Zap extension
then add a zap extension for channel 1 in AMP and if it shows your Zap
trunk is channel 2 you should create a zap trunk for channel 2 in AMP.

Once this is done, reboot your PC and when Asterisk starts, use AMP
to add a route for incoming calls or asterisk will not answer your
trunk. Similarly, to make outbound calls you will need an outbound
route. Set them up as per setting up routes in the earlier chapters of
this document.

If you have this card installed, you will need to edit the
following files; zapata.conf and zaptel.conf as per the X100P card in
the previous section.

9.2.2 modules.conf (modprobe.conf for AAH 2.x)
You will need to edit the modules.conf, or modprobe.conf to add the necessary option for usage in Australia.

The example below is for AAH 1.x. where you need to add the following line;

options wctdm opermode=AUSTRALIA fxshonormode=1 bootstringer=1

Your modules.conf (AAH 1.x) should look like the example below:

alias eth0 e100

alias sound-slot-0 es1370

post-install sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -L >/dev/null 2>&1 || :

pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null 2>&1 || :

alias usb-controller usb-uhci

alias char-major-196 torisa

options wctdm opermode=AUSTRALIA fxshonormode=1 boostringer=1

options torisa base=0xd0000

post-install tor2 /sbin/ztcfg

post-install torisa /sbin/ztcfg

post-install wcusb /sbin/ztcfg

post-install wcfxo /sbin/ztcfg

post-install wctdm /sbin/ztcfg

post-install ztdynamic /sbin/ztcfg

You will only need to add the line in red. Do not change anything else.

Or, in AAH Ver.2.1, you may also do the following:

Locate the line 'install wctdm /sbin/ztcfg-- --ignore-install wctdm
&& /sbin/ztcfg' and edit it to reflect the following:

install wctdm opermode=AUSTRALIA fxshonormode=1 boostringer=1 /sbin/ztcfg-- --ignore-install wctdm && /sbin/ztcfg

Note: as of Zaptel
Drivers 1.2.4, by selecting opermode=AUSTRALIA the zaptel drivers
automatically add the 'boostringer=1 , fxshonormode=1'

Also see Appendix E.3 (Users Suggestions)

9.3 REBUILDING ZAPTEL DRIVER
Every time there is a kernel update with yum (which is the case
with Asterisk and CentOS), ZAP device support needs to be rebuilt using
the new kernel. Unfortunately, this will cause a slight problem as
RedHat bug caused the rebuilding process to fail.

The following is the fix - source Nerd Vittles http://nerdvittles.com/index.php?p=123

Log into your new server as root and issue the following commands:

cd /usr/src/kernels/2.6.9-34.EL-i686/ include/linux

mv spinlock.h spinlock.h.old

wget http://nerdvittles.com/aah27/spinlock.h

Once the file has been retrieved, reboot using the following command:

shutdown -r now

When the reboot completes, you can start rebuilding the support for
your ZAP devices or for that matter, ztdummy if you don’t have any ZAP
devices.

Log in as root and type the following command:

rebuild_zaptel

Then reboot your system:

shutdown -r now

Now log in as root again and enter the following command:

amportal stop

genzaptelconf

Reboot once again:

shutdown -r now

..and you're done.

(See also user Users’ Suggestions)

9.4 SIPURA SPA3000 AS A PSTN INTERFACE
To those new to the SPA3000, there is a simplified installation
and configuration instruction by JMG Technology. While it is directed
mainly at standalone ATA users, it gives a good insight of the Sipura
SPA3000’s capabilities.

I have come across a few people in the various forums wanting to
use their Sipura SPA-3000s as FXO front-end to their Asterisk@Home
boxes. To help them in their endeavours, I've put the following
together, as no one single source of information that I've found so far
has a config that would actually work for me.

9.4.1 Log in to SPA3000
Login to your SPA-3000 as admin/advanced.

Before you change anything, I'd suggest taking a snapshot (i.e.
just save the .html page) of your current SPA-3000 configuration, just
in case you ever need to refer back to your own customisations.

If you're not already running the latest SPA-3000 firmware, then
upgrade it to the latest version (at the time of writing, it's 3.1.5a).
Take another snapshot for good measure. Nothing should have changed in
your settings, except that you have a few extra options that you didn't
have before.

Now reset SPA-3000 back to factory defaults, because I'm only going
to list the minimum changes required to keep things simple. Take
another snapshot now too, in case you ever want to know what the
defaults were.

9.4.2 Change the settings
System tab

DHCP: No

Static IP: something on your local subnet e.g.; 192.168.1.200

NetMask: 255.255.255.0

Gateway: your router's IP address e.g.; 192.168.1.254

Primary DNS: your ISP's primary DNS address e.g.; 203.12.160.35

Secondary DNS: your ISP's secondary DNS address e.g.; 203.12.160.36

Regional tab

Dial Tone: 400@-19,425@-19;10(*/0/1+2)

Busy Tone: 425@-10;10(.4/.4/1)

Reorder Tone: 425@-10;10(.2/.2/1)

Ring Back Tone: 400@-19,425@-19,450@-19;*(.4/.2/1+2+3,.4/.2/1+2+3,0/2/0)

Ring 1 Cadence: 60(1.5/3.4)

Ring 3 Cadence:
60(1.5/3.4,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2)

CWT8 Cadence: 30(.2/.2,.2/4.4)

Hook Flash Timer Min: .07

Hook Flash Timer Max: .13

Delete all the Vertical Service Activation Codes.

FXS Port Impedance: 220+820||120nF

Line 1 tab

Proxy: IP address of your Asterisk box e.g.; 192.168.1.234

Register Expires: 60

Display Name: Whatever

User ID: Asterisk extension number e.g.; 200

Password: password for that extension

Silence Threshold: medium

DTMF Tx Method: INFO

Hook Flash Tx Method: INFO

Dial Plan: (*xx|000|0011xxxxxxxxxxx.|0[23478]xxxxxxxx|09xxxxxx|1100
|122[135]|1222xxxxxxx|12510[12]|12554|1[38]00xxxxxx|13[1-9]xxx
|1747xxxxxxx|2xx|393xxxxxx|3xxxx. |[4689]xxxxxxx|7777|899060xxxxx.) for
example

(*xx.|x.) will work, but I like to do a bit of sanity checking, etc.

PSTN Line tab (method 1)

Proxy: IP address of your Asterisk box e.g.; 192.168.1.234

Register: no

Make Call Without Reg: yes

Ans Call Without Reg: yes

Display Name: No name

User ID: PSTN

Password: password

Silence Supp Enable: no

Echo Canc Enable: no

Echo Canc Adapt Enable: no

Echo Supp Enable: no

FAX CED Detect Enable: yes

FAX CNG Detect Enable: yes

FAX Passthru Codec: G711u

FAX Codec Symmetric: no

FAX Passthru Method: None

DTMF Tx Method: INFO

FAX Process NSE: no

Dial Plan 1: (S0<:T0298765432>) for example

VoIP Caller Default DP: none

PSTN Ring Thru Line 1: no

PSTN CID For VoIP CID: yes

PSTN Answer Delay: 2

PSTN Ring Thru Delay: 3

PSTN Ring Timeout: 4

PSTN Hook Flash Len: .1

Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2)

FXO Port Impedance: 220+820||120nF

On-Hook Speed: 26ms (Australia)

(Source reference: Colin Swan)

Or alternatively you may want to adopt the second method for the PSTN Line Tab, which I am currently using.

PSTN Line tab (method 2)

Proxy: IP address of your Asterisk box e.g.; 192.168.1.234

Register: no

Make Call Without Reg: yes

Ans Call Without Reg: yes

Display Name: No name

User ID: PSTN

Password: password

Silence Supp Enable: no

Echo Canc Enable: no

Echo Canc Adapt Enable: no

Echo Supp Enable: no

FAX CED Detect Enable: yes

FAX CNG Detect Enable: yes

FAX Passthru Codec: G711u

FAX Codec Symmetric: no

FAX Passthru Method: None

DTMF Tx Method: INFO

FAX Process NSE: no

Dial Plan 1: (S0<:s@YourAsteriskIP>) e.g. (S0<:s@192.168.0.101:5060>)or try w/o the port designation

VoIP Caller Default DP: none

PSTN Ring Thru Line 1: no

PSTN CID For VoIP CID: yes

PSTN Answer Delay: 2

PSTN Ring Thru Delay: 3

PSTN Ring Timeout: 4

PSTN Hook Flash Len: .1

Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2)

FXO Port Impedance: 220+820||120nF

On-Hook Speed: 26ms (Australia)

Using this alternative method, you will not need to create an
Inbound Route for this channel as the call is sent directly to your “s‿
extension as defined in your incoming call setting. You may also get
CLID if your Telco has activated incoming Caller ID on your phone.

User 1 tab

Default Ring: 3

Default CWT: 8

9.4.3 Add SIP Trunk
Then in AMP, add a SIP trunk.

Outbound Caller ID: <0298765432> (for example)

Maximum Channels: 1

Dial Rules: 0+NXXXXXXXX (for example)

0011+ZXXXXXXXXXX.

Trunk Name: telstra (for example)

Peer Details:

canreinvite=no

context=from-pstn

host=the IP address of your SPA-3000 (for example; 192.168.1.200)

insecure=very

nat=no

port=5061 (for example)

qualify=yes

secret=password

type=peer

username=PSTN

User Context: telstra-incoming (for example)

User Details:

canreinvite=no

context=from-pstn

host=the IP address of your SPA-3000 (for example; 192.168.1.200)

insecure=very

nat=no

port=5061 for example

secret=password

type=user

username=PSTN

Leave "Register String" empty

Then add a DID Route of T0298765432 (for example), which goes to your chosen Destination. (Source reference: Colin Swan)

See the alternative configuration that I am currently using for the PSTN Tab in Notes

Also see Eliminating echo problems in Appendix E.4 in Sipura SPA-3000

Is Voip for You?

Is VoIP for You?

Whether VOIP is for you or not rely on a number of or combination
of factors. Some economic and quality considerations should be
examined.

What is it going to cost?

Assuming that you already have a broadband service, a router, and a Windows PC to run the softphone, the cost will be minimal.

If you already have a spare computer to dedicate to this task, then
the cost is almost nothing unless you need to buy an audio headset
($15.00 from Dick Smith - Australia.) for the softphone. If you do not have a spare
PC with the above specification, then you may be able to buy one from
your local swap meets for under $200.00, which may include a monitor.
Ensure that the PC has an Ethernet NIC for connecting to your home
network.

Your only other initial cost will be the $20.00 or so activation
fee to Oztel (or other VSP of your choice), if you want the ability to
make PSTN calls. If you want to restrict all your calls to VOIP only,
it may not cost you anything at all.

Some VSPs like Pennytel, Astratel, Spantalk etc will register you
for SIP communication for free provided that you do not need to make
PSTN calls.

All these “Major Expenses" will be recovered when you receive your monthly Telstra or Optus phone bills.

What will the Quality of the phone calls be?

If you are expecting the quality to be as good as your existing
PSTN calls, you will be somewhat disappointed, but if you will be happy
with a quality that is not quite but close to your existing PSTN calls,
you might be in luck.

VOIP via the Public Internet is very much dependant on a number of
factors – available bandwidth not withstanding, your usage habit of the
internet and LAN traffic and equipment quality, amongst others, also
play very important roles.

Linux CLI Commands

Entering the Asterisk Console
asterisk -r

Checking Current System Load
top

Interrupt Information
cat /proc/interrupts

RAID Array Information
cat /proc/mdstat

Checking the Routing table
netstat -rn OR route

Checking CPU Information
cat /proc/interrupts

Checking Memory Information
cat /proc/meminfo

Running tcpdump
tcpdump -A -s 10000 port <port> and host <host>

Running PING tests
ping -i 0.02 -c 500 -s 270 <host>

Intensive Performance Information
vmstat 1

Current Wanpipe Version
wanrouter version

Current system processes
ps aux

Current Networking Information
ifconfig -a

Duplexing Diagnostics
mii-tool

Rsync Usage
rsync -av -essh /path/to/file <remote_site>:/path/to/file

SCP Usage
scp /path/to/file <remote_host>:/path/to/file

Checking Disk Space
df -h

Setting up Phones

Some docs on how to setup up hard and soft phones with freepbx

Setting Up Voicemail

Setting up Voicemail

Voicemailboxes are typically created when used for the first time.

Assigning Voicemail PasswordsYou must enter a voicemail password when creating an extension enabled with voicemail. Choose a password that is at least 4 digits. Resist the impulse to standardize the default. Most people won't change it, and the system will be insecure.
Instructing New Users

New users should, as matter of policy, be sent an email with both general instructions, their voicemail initial password, and directions to change it.

Voicemail in email feature

In order for these emails to pass through spam filters, the following are recommended:

1. Set your hostname to be a fully-qualified domain name. Use dyndns.org if you don't want to pay for a real one.
2. Don't send from dynamic IP addresses.

Voicemail pager feature

This gives a short description of the message envelope, suitable for emailing to a wireless carrier's email gateway.

Voicemail Locator (VMX) Feature

This optional feature lets users set up a short menu before voicemail takes the actual message. Common options are 'press 1 for my cell, 2 for my assistant or just leave a message after the tone.' 

Resetting

It's commonplace to have to reset the passwords as people leave the company.

System Tools

Area for additional system tools for Asterisk and FreePBX

Module Admin - Read This First!

Overview
Everything in FreePBX is a module, so you need to enable all the modules you want to use. If you do enable something you don't use, it won't matter - it'll just make clicking the red 'Reload Bar' take a little longer, as it goes through and asks every modules 'What would you like done?. If you're using a lower powered system (Say, a Piii 500 or slower) you might want to be a bit selective with the modules you use, but it will _only_ affect the speed of FreePBX, not the speed of Asterisk itself.

Modules and their descriptions

1. Core: This coveres your basic 'Extensions' and 'Trunks' etc. You pretty much always want to have this enabled
2. Feature Code Admin: For configuration of 'call features', such as DND and Call Forwarding.
3. Follow Me: Provides a 'Follow Me' service
4. Misc Destinations: Allows you to use any number you can dial as a destination
5. PHP Info: A sample module
6. Ring Groups: Lets you define a group of extensions (or external devices) to be called when a certain extension is rung.
7. Time Conditions: Lets you define a particular time period and alternative destinations based on whether you are current in the time period specified. You then use the Time Condition itself as a destination in other locations.
8. Call Forward: A call feature code.
9. Call Waiting: A call feature code.
10. Do-Not-Disturb: A call feature code.
11. Online Support: Enables you to contact developers and other online, immediately.
12. nfo Services: A call feature code.
13. Voicemail: A call feature code
14. IVR: Lets you create IVR (i.e. Digital Receptionist) menus
15. On Hold Music: Lets you define Music On Hold categories and upload MP3s to use for each category.
16. PIN Sets: Lets you check a range of PIN's. Currently only used for Trunks.
17. Paging and Intercom: Lets you define paging groups (intercom not currently supported) to automatically page a group of extensions.
18. Queues: Lets you create call queues
19. Recordings: Lets you create Recordings that can be used in various places (like Digital Receptionists or Queues)
20. DISA: Lets you create DISA (Direct Inward System Access) destinations (only available in the SVN trunk version)
21. Asterisk CLI: Adds a tool that allows you to issue commands to the Asterisk CLI interface
22. Conferences: Lets you create MeetMe conferences.
23. Backup & Restore: Adds a tool that allows you to backup or restore your freePBX configuration

Administrators

This module lets you limit the sections of freePBX to certain users.

Configuration

This module may not be active by default, and will say 'NOTE -
AUTHTYPE is not set to 'database' in /etc/amportal.conf - Module
crippled'. To enable it, you need to change AUTHTYPE in
/etc/amportal.conf to 'database'. If you have already tried to add
users before changing AUTHTYPE to 'database', delete them now and read
the "important warning" below.

Trixbox or Asterisk@Home users have some extra work due to the
Apache level authentication with the "maint" user. You will have to
comment out or delete the following lines in
/etc/trixbox/httpdconf/trixbox.conf for Trixbox or
/etc/httpd/conf/httpd.conf for A@H for this to work:

Password protect /var/www/html/admin

<Directory /var/www/html/admin>

AuthType Basic

AuthName "Restricted Area"

AuthUserFile /usr/local/apache/passwd/wwwpasswd

Require user wwwadmin maint

</Directory>

Then you have to restart httpd (/etc/init.d/httpd restart) and
possibly amportal (amportal restart) too. You should now be able to
login with admin/admin and create/change users in the Administrators
module. Be sure to change the admin password straight away!

Important Warning! Read before using this module

It is quite easy to lock yourself out of freePBX if you enable
AUTHTYPE after you have added users. Don't do it. If you want to use
this, then you must have NO USERS CREATED before you turn it on. If you
don't heed this advice, and turn it on with an existing user there,
then nothing you type in will let you access freePBX. You'll have to
turn it back off, and then delete any existing users.

Adding a User

Enter a username and password in the General section. If this is
the first user, make sure that you select 'ALL SECTIONS' in the 'Admin
Access' list so you can get back in there. As soon as you add the first
user, you will then be prompted for a username and password. Log in as
the user you've just created.

Limiting User Access

• Department Name
  • Have been unable to find any documentation on this . Help?
• Extension Range
  • When this user is logged in, they will only see the range
    specified here. This is useful if you're setting up multiple tenants on
    the one system.
• Admin Access
  • This is a multiple-selection box. You can select a range of
    areas they're allowed to access by either holding down Control (or
    'apple' on Mac's) and selecting individual ones, or dragging the mouse
    over the list of ones you want to give them access to.

Announcements

The Announcement module alows you to play an announcement to a caller. You can then send the caller to another destination or back to the IVR that sent him to the announcement.

For example:

1. IVR: "Hello, thank you for calling the North Carolina School district. To hear the latest updates about changes to thew school schedule due to the current weather conditions, press 1." After pressing 1, the caller will then hear the announcement regarding todays schedule. Then the caller will be returned to the IVR where they can press another option (2 for the school administration, etc.)
2. You can play a message to the caller after the select an IVR destination - but befor the call is transfer. I.e. if the caller pressed 1 for sales, you could play "transfering to the sales team" - and then send the caller to the sales queue.

Configuration

• Description: This is the name of the announcement.
• Recording: This is the file to be played. You can add more recording using the System Recordings module.
• Repeat: This option sets a repeat key.If the caller presses this key the announcement will be replayed.
• Allow Skip: Alows the caller to press any key, to skip the announcement.
• Return to IVR: Returns the caller to the IVR after they have heard the announcement
• Don't Answer Channel: Check this to keep the channel from explicitly being answered. When checked, the message will be played and if the channel is not already answered it will be delivered as early media if the channel supports that. When not checked, the channel is answered followed by a 1 second delay. When using an annoucement from an IVR or other sources that have already answered the channel, that 1 second delay may not be desired.

Asterisk CLI

Asterisk CLI command

This allows you to run a command as if it was typed into the asterisk CLI.

Examples

* sip show peers

o This displays all the known SIP devices, and their state, according to Asterisk

* show channels

o Show any channels that are in use at the moment

* soft hangup Zap/1

o Hangs up the Zap/1 channel

Asterisk Info Module

Asterisk Information Module - Gives the use a snapshot of the system state such as SIP channels (calls in progress) SIP peers (SIP phones, VoIP trunck) etc...

This module will give you a summary of the following.

Summary: Summary of your system state.
-system uptime
-active SIP channels
-active IAX2 channels
-SIP registration
-IAX2 registration
-SIP Peers
-IAX2 peers

Channels: Summary of the systems channel information:
-Active Channels
-SIP channels
-IAX2 channels

Peers: Summary of the systems Peers:
-SIP peers
-IAX2 peers

SIP info - summary of the systems sip information:
-SIP registry
-SIP peers

IAX2 info - summary of the systems IAX2 information
-IAX2 registry
-IAX2 peers

Conferences - summary of the active conferences / MeetMe.

Subscriptions / Notify - registered Asterisk dial plan Hints.

Voicemail Users - A list of the extensions that have VM enabled and the number of new Voicemails.

Full Report: Gives you a full report of all the above information in one report.

Refresh - hitting this button will refresh the snapshot of the Asterisk Information.

Asterisk Logs

No documentation availabe. You can help by contributing to this book pate.

Asterisk Manager API Settings

No documentation availabe. You can help by contributing to this book pate.

Backup and Restore

Overview

You can configure a regular backup schedule to ensure that you
have a copy of your Asterisk and freePBX configuration, voicemail and
CDR records. You can also restore a previous backup, in case of data
loss or a major configuration fault. Backups are stored on the file
system at /var/lib/asterisk/backups. You should make a point of making
an offline copy of important backups.

Add Backup Schedule

• Schedule Name: Give this backup a friendly name (e.g.
  "Daily" or "Voicemail") to accurately identify what you're backing up.
  This will make future restores easier.
• Voicemail: Enable this if you want to include voicemail
  messages in this backup. This could seriously increase the size of your
  backups because you are backing up potentially large audio files.
• System Recordings: Enable this if you want to backup custom
  System Recordings you may have created for a Digital Receptionist or
  Queue. Again, this could increate the size of your backups because of
  the size of some audio files.
• System Configuration: Enable this option to backup your
  Asterisk and freePBX configuration data, including the MySQL and
  Asterisk databases. We recommend this be enabled for all backups.
• CDR: Enable this option to backup your Call Detail Records.
• Operator Panel: Enable this option to backup the Flash Operator Panel configuration.

Run Schedule

You can chose a pre-configured schedule from the drop-down, or
configure your own schedule using the Minutes, Hours, Days, Months and
Weekdays select boxes. The pre-configured options are: Daily (at
Midnight), Weekly (on Sunday at Midnight), Monthly (on the 1st at
Midnight) or Yearly (on the 1st January at Midnight).

Restore from Backup

This will list all the backups that are currently on your system
(located at /var/lib/asterisk/backups). Click on the backup you wish to
restore.

Upgrading from A@H:

Upgrading from an older version of A@H to freePBX causes restores to STOP WORKING. You need to totally clean out your /var/www/html/admin directory first, then re-run install_amp.

cd /usr/src/freepbx

rm -rf /var/www/html/admin

./install_amp

*** NOTE ***
I have tested a revision of this. Restoring an old version of FreePBX (2.2.X)to the newest version of FreePBX 2.4.X.

1. install your new system with FreePBX 2.4.0
2. install the Backup / Restore module
3. restore your old backup (I always restored everything for testing)
4. SSH into /usr/src/tbm-pbxconfig-5.0.0
5. run ./install_amp *this will fix all the web interface problems and the modules that were hosed.
5. Do not hit the orange bar yet.
6. Update all the modules, check your extensions then hit the orange bar and reload.

the system should be close to perfect.

This has only been tested. Long term testing has not been done.

Blacklist

This module allows you to manage the blacklist in astDB thru the FreePBX web interface.

Add or replace entry:

Number: Enter the number you want to block.

Adding a number to the blacklist will not allow the number into your system. The blacklisted caller will hear "The number you have reached is not is service."

Hit *32 to blacklist the last number called into your system.

On some phones you can setup a speeddial to blacklist callers.
speeddial *32.

Callback

The Callback module allows you to setup a destination that calls a user back and provides them access to an application. An example of this would be a caller that dials your system, disconnects, and is called back and then provided a DISA
dial-tone to make a phone call.

Add Callback

• Callback Description: This is the name/description of the callback
• Callback Number: This is the number to be called
• Delay Before Callback: Optionally, you can enter an amount of time that the system should wait before placing the call

Known Issues

Some issues have been encountered with this plugin:

1. Callbacks frequently fail, as it appears Asterisk disconnects
   before the associated callback PHP script (located in
   /var/lib/asterisk/bin) may complete the callback transaction.
2. If one uses the 'CLI' callback capability, then you may
   encounter issues with being able to actually dial people back. The
   reason for this is that the telco only provides a 10-digit CLI, and if
   your provider requires you to add a '1' there is no way to do this
   without hacking the plug-in. Further, if your own dialplan requires an
   access code like '9' you also encounter this problem.
3. There is no way to provide a number of retries, retry time or
   wait times for an answer. This app will try once, and if it fails, will
   not try again.

Alternative Solution

A replacement script, written in Ruby, has been created to replace
the /var/lib/asterisk/bin/callback PHP script provided with the
callback plugin. The Ruby script takes a different approach to
providing the callback capability, while still supported the HTML form
provided with the plug-in with FreePBX. The enhancements are:

- Provides the ability to set the access code to prepend to the recognized CLI number for a callback

- Uses call files instead of the Manager API

- Allows one to set the number of retries, the retry wait time and the wait time for a caller to pickup an attempt

To use the script below do the following:

(Note: This script requires that Ruby 1.8.x be installed on your Asterisk/FreePBX system)

1. Move /var/lib/asterisk/bin/callback to /var/lib/asterisk/bin/callback.original (to back it up)

2. Download the attached script and copy to
/var/lib/asterisk/bin/callback and then do a

chmod +x
/var/lib/asterisk/bin/callback

3. Add these options to the end of your configuration file at /etc/amportal.conf:

CALLBACK_PREFIX=91
CALLBACK_CLI=Callback
CALLBACK_RETRY=1
CALLBACK_INTERVAL=10
CALLBACK_WAIT=30

Caller ID Lookup Sources

This module provides the ability to specify Sources where inbound calls can have their Caller ID looked up so Caller ID Names can be used or changed.

The Caller ID Lookup Sources module enables your FreePBX system to lookup Caller Names that are related to your number whether they be in your phonebook, in a database, or via an HTTP lookup. The module can also be used with scripts in the agi-bin directory of your asterisk configuration.

------------------------------------------------------------------
Add Source

A Lookup Source let you specify a source for resolving numeric caller IDs of incoming calls, you can then link an Inbound route to a specific CID source. This way you will have more detailed CDR reports with information’s taken directly from your CRM. You can also install the phonebook module to have a small number <-> name association. Pay attention, name lookup may slow down your PBX.

Source Description: A description for this source.

Source Type: Choose the source type.
Internal: uses astDB as a lookup source. Use the phonebook module to populate it.

ENUM: Uses DNS to look up the callers Name; it uses ENUM lookup zones as configured in enum.conf

HTTP: This executes an HTTP GET passing the caller number as an argument to retrieve the correct name.

MySQL: the queries a MySQL database to retrieve the caller name.

Cache results - Decide whether or not to cache the results to a astDB; it will overwrite present values. It does not affect internal source behavior.

----------------------------------------------------

HTTP Lookup Configuration:

Configuring the module to lookup a Caller Name (cn) via http lookup is simple. Most http lookup providers will provide you with a string you'll need to query with (query string). We'll need to break up the string into it's various components to populate the CID Lookup Source Fields.

For the purposes of this tutorial, i'm going to use configuring metrostat as an http provider. They expect queries in the form: http://cnam1.edicentral.net/getcnam?q=C&f=S&dn=[NUMBER] where [NUMBER] is a 10 digit telephone number.

Source Description: Metrostat
Source Type: HTTP
Cache Results: (leave this unchecked)

Host: cnam1.edicentral.net
Port: (leave blank)
Username: (leave blank)
Password: (leave blank)
Path: /getcnam
Query: ?q=C&f=S&dn=[NUMBER]

ok.. so let's break this up so that we can see what we did. Source Description is nothing more than a name you want to call the lookup source. Since it's metrostat, we're just going to call it metrostat.

Source Type: Here, we can specify whatever the lookup source type that is supported for the query. In this particular case, it's an http lookup so we're going to specify http.

Cache Results: You can check this and the system will keep successful lookups for future use, thus eliminating excessive remote lookups for numbers that have already called you in the past.

Host: This is the hostname or IP address of the host you'll be querying. In our example, it's cnam1.edicentral.net.

Port: You can pass your query to any port that is running httpd. This is usually left blank if it's on the standard port (80). In our example, we're leaving it blank, because it's on the standard http port.

Username and Password fields: The username you're issued by the provider if it's needed, otherwise, it's blank. In this case, it's blank. This goes for the password as well.

Path: This is the first thing after the hostname (including the slash. Up to, but NOT including the '?' symbol. That's reserved for the query line otherwise it just won't work.

Query: This is everything after the Path entry. In our example, we had '/getcnam' so we'd be putting the following in here. '?q=C&f=S&dn=[NUMBER]'. The [NUMBER] variable will automatically be filled in with the incoming caller number.

After you've filled out and submitted your changes (remember to hit apply changes), go to the incoming route and select the appropriate caller id lookup source, submit the changes, and apply and you're good to go! enjoy!

Conferences (MeetMe)

Information

Conferences is a standard multi-party conferencing facility that is available as a destination.

Conference Details

• Conference Number
  • This is a number that local users can dial to join the conference
• Conference Name
  • This is used as an Identifier, along with the number, when picking a conference as a destination
• User PIN and Admin PIN
  • If either of these options are set, anyone calling into the
    conference will be prompted for a PIN. If 'user' is left blank, they
    can just push '#' to enter. The only use of 'Admin' is to not actually
    open the conference until the admin user has arrived. If 'Music On
    Hold' is enabled, users will be placed on hold with the 'default' Music
    On Hold class.

Conference Options

• Join Message
  • This is a sound that is played to all users upon entering.
• Leader Wait
  • When there is an Admin PIN set, the conference won't start until the 'Admin' user joins. See above.
• Quiet Mode
  • Usually a 'bing' noise is played when a user enter or leaves
    the conference, alerting other members to the fact that someone has
    joined or left. You can disable that by selecting 'Yes' here.
• User Count
  • When someone joins, the conference will say 'There are (number) people in this conference"
• User Join/Leave
  • When someone connects to the conference, it will ask them to
    record their name. The conference will then announce when they join and
    leave, by name.
• Music On Hold
  • Totally enables or disables Music on Hold in this conference.
• Allow Menu
  • Enables the user or admin to enter an the management mode by pushing '*'. The commands whilst in management mode are:
    • 1: Mute yourself
    • 4 or 6: Decrease or Increase the Conference Volume (eg, the sound you hear)
    • 7 or 9: Decrease or Increase your Volume (eg, the sound other people hear)
  • Additionally, Admin users have the added features of:
    • 2: Lock or Unlock the conference
    • 3: Eject the last person that called
• Record Conference:
  Toggle this to yes if you would like to record the conference.

Core Module

Description

The Core Module is a collection of the fundamental modules that make up FreepPBX. It includes:

Extensions

Used to create User Extensions

FreePBX User

Used to create a FreePBX User (when operating in Devices and Users Mode) *This module is not present in the latest version of FreePBX 2.4.x

Devices

Used to create a FreePBX Device (when operating in Devices and Users Mode) *This module is not present in the latest version of FreePBX 2.4.x

Inbound Routes

Used to route incoming numbers, DIDs, to various destinations

Outbound Routes

Used to route outbound dial patterns to various Trunks

Trunks

Used to configure Zap, SIP, IAX and other types of Trunks to the PSTN or to interconnect with other PBXs

Administrators

Used to create FreePBX Administrators and provide access to limitted modules or extnension ranges *note this module will not work and the new users will not be able to log into the system unless AUTHTYPE is set to 'database' in /etc/amportal.conf

General Settings

A collection of system wide and defaut settings used by different systems within the PBX.

Custom Extension and Destinations

Custom Extension and Destinations
This documentation needs writing, for adding custom extensions and destinations to the registries. Contact us if you would like to write this page.

As a place holder until someone wants to update this page, here is a good writeup on using destination:

http://ronaldgibson.googlepages.com/unknowndestinationfreepbxv2.4.0beta

Customer Database Module

No documentation availabe. You can help by contributing to this book page.

Day / Night Mode Control

This module alows for the inbound routing changes to be placed via a phone call - without requiring access to the web configuration page. The module is useful for any situation where inbound routing needs to be changed by end users on-the-fly. An example of this would be a situation where an office would close early because of bad weather. In this case, just dial the appropriate feature code, and voila! all inbound calls go straight to the "night" IVR (or voice mail, or calls get forwarded to an answering service, etc.)

Configuration

• Day/Night Feature Code Index: This is just a descriptive number to help identify which this particular feature code
• Description: A descriptive name for this feature code
• Current Mode: The mode that the feature code is currently set to (is now "day" or "night"?)
• Optional Password: A password that protects the current state (dayor night) from being toggled by unauthorized personnel.

For an in depth tutorial, click here.

Destinations

Destinations are provided by various modules. If you don't have
these modules installed, you will not see these destinations. Current
modules are:

Core

Provides the ability to send a call to an Extension, or an extensions Voice Mail box directly.

Queues

Lets you send a call into a Queue, which is an intelligent Ring Group.

IVR

This lets you jump to a specific Digital Receptionist IVR.

Time Conditions

This lets you do the equivalent of an 'IF', and you can pick two
different destinations depending on the time of the call. See Time
Conditions for more information.

Ring Groups

A Ring Group is a group of phones that ring simultaneously or sequentially. A dumber version of Queues

Conferences

Lets you use a standard tele-conferencing application. Please read Conferences before using

Misc Destinations

This lets you enter a number, just as if you dialed it on one of
your phones. Useful for making an "outside" number (such as a mobile
phone) a destination.

Custom App

You can create your own application using the standard Asterisk
Dialplan language in the file /etc/asterisk/extensions_custom.conf, and
reference it with this. An example would be:

custom-count2four

exten => s,1,SayDigits(1234)

exten => s,2,Hangup

After placing this in /etc/asterisk/extensions_custom.conf, you can send callers to this by using 'custom-count2four,s,1'

Dictate Module

The Dictate Module will not be found in the list of modules. When loaded it adds a dictation area to the Extension
Module. Load the module then look in an extension for 'Dictation Services'

Dictation Services

• Dictation Service:
  Enable or Disable with this drop down menu.
• Dictation Format:
  This is the sound format that your dictation is saved in. Choose from: Ogg Verbis, GSM, or WAV.
• Email Address:
  This is the email address that the completed dictations are sent to.

To dictate enter *34 on the phone. Enter a file name with numbers followed by the pound key, for example 1234#.
To start dictate dial 1 to switch to record mode then dial * to start recording. If you want to start over, dial 8.
Toggle between pause and recording by dial *.
When you are done with the recording, dial 1 to go to playback mode then dial * to start to listen to your dictation, press * to toggle pause.
Dial 2 to toggle fast playback. Dial 7 or 8 to seek forward and reverse.

To mail the dictation, dial *35 then enter your file name used previously.

DISA

Overview

DISA (which stands for Direct Inward System Access) allows you to
provide an internal dialtone to external callers. When you configure a
DISA destination, you can use it as a menu destination within a Digital
Receptionist, so that you can get an internal Asterisk dialtone. This
means you could call into your Asterisk system and dial out as if you
were using an extension connected to the Asterisk box itself.

The security implications of DISA are obviuous! Make sure you have a proper authentication scheme in place so that unautherized callers cannot abuse your system!

Changelog

V2.0 - Added Response Timeout, Digit Timeout and Reuqire Confirmation

Configuration

• DISA Name: A friendly name to help you identify the DISA destination.
• PIN: A code required to be input by the
  remote user to access the dialtone. You should always require a PIN for
  security purposes! You can, if you must, leave it blank. The voice
  prompt will still say 'Enter Password', but you won't need to. This is
  on the list of minor annoyances that need to be fixed.
• Caller ID: You can change the outgoing caller ID for this DISA by setting an override value.
• Response Timeout: The maxium amount of time it will wait before hanging up if the user has dialed an incomplete or invalid extension
• Digit Timeout: The maximum time delay between digits
• Require Confirmation: Before prompting for
  password, it will say 'Press 1' every 3 seconds until a digit
  (anything, it doesn't have to be '1') is pressed. It will then continue
  on and ask for a password. This is mainly useful if dialling out
  through a device (eg, X100P, TDM400, or a poorly configured VSP) that
  doesn't tell you when the call is actually answered.
• Context: By default, internal context
  (from-internal) is used. You could provide a custom context to limit
  the access for this DISA. (Only for the experienced)
• Allow Hangup:When checked allows the current call to be ended and return the
  caller to the dial tone by pressing the Hangup feature code. (**)

DUNDi Lookup and Extension Registry Proxy

DUNDi Lookup and Extension Registry Proxy
This module provides a proxy for the extension registry feature in FreePBX. If you have a DUNDi trunk configured in FreePBX to other branch offices, and a route defined to access it, then this module will proxy and check for extension number duplication in other branch offices when creating new extensions, ringgroups or any other extension that can be dialed.
The module does not consider the outbound dialing rules. It simply checks the DUNDi cloud down all configured DUNDi routes, and if another system indicates they have that extension, then it will consider this a conflict and report it back as such since a change in routing rules would easily expose this conflict.
As a secondary function, the module allows you to easily check for numbers present within the configured DUNDi routes in a simple GUI page.

Extensions

Information

This area is for handsets, softphones, paging systems, or anything
else that could be considered an 'extension' in the classical PBX
context.

Overview

Defining and editing extensions is probably the most common task
performed by a PBX administrator, and as such, you'll find you'll
become very familiar with this page. There are presently four types of
devices supported - SIP, IAX2, ZAP and 'Custom'. This page also
configures how voicemail is handled on a per-extension basis.

Adding a new extension

Phone Protocol

Pick one of SIP, IAX, ZAP or Custom

• SIP is the Standard protocol for VoIP handsets and ATA's. The Session Initiation Protocol (SIP) is a signalling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet.
• IAX is 'Inter Asterisk Protocol', a newer protocol supported by only a few devices (eg, PA1688 based phones, and the IAXy ATA)
• ZAP is a hardware device connected to your Asterisk machine - Eg, a TDM400, TE110P.
• Custom is a 'catch all', for any non standard device, eg
  H323. It can also be used for "mapping" an extension to an "outside"
  number. For example, to route extension 211 to 1-800-555-1212, you
  could create a custom extension 211 and in the "dial" text box you
  could enter: Local/18005551212@outbound-allroutes

Extension Number

This must be unique. This is the number that can be dialled from any other extension, or directly from the Digital Receptionist if enabled. This

may be any length, but conventionally a three or four digit extension is used.

CID Num Alias

The CID Number to use for internal calls, if different from
the extension number. This is used to masquerade as a different user. A
common example is a team of support people who would like their internal
callerid to display the general support number (a ringgroup or queue).
There will be no effect on external calls.

SIP Alias

If you want to support direct sip dialing of users internally
or through anonymous sip calls, you can supply a friendly name that can
be used in addition to the users extension to call them.

Secret, aka 'Extension Password'

This is the password used by the telephony device to authenticate
to the Asterisk server. This is usually configured by the administrator
before giving the phone to the user, and is usually not required to be
known by the user. If the user is using a soft-phone, then they'll need
to know this password to configure their software.

NAT

very important to set this to 'yes' if you want to be able to use
your phone behind a nat firewall seperate from the lan that your
asterisk box is on.

Disallow

disallow any codec you do not want to use. common setting for this
if you want to make sure a device only uses the codec set in the allow
section is "all" (without the quotes).

Allow

allow a codec of your choice. primarily as an example i would use
"g729" or "GSM" (without the quotes), only one codec can be set here.
This is usefull if used with the disallow option set to "all" and you
set the definitive codec you want to use on allow, garunteing that you
will use that codec.

Mailbox

This option allows you to set what mailbox you would like to use.
Normally you would use 'your extension'@device . but if you want to say
have an extension's phone's light or dial tone indicate when a
different box has voicemail you can set it to 'extension vm'@device. so
say extension 1002 wants to know when 1001 has vm, then set this to
1001@device on 1002's mailbox setting.

Record Incoming

Option to record the calls received on this extension. There are three options:

• Always
• Never
• On-Demand (User can dial '*1' to enable whilst in a call)

Record Outgoing

Same as above but for outgoing calls

Voicemail and Directory

Selecting 'Disabled' turns off voicemail for this extension totally, and these further options are hidden.

Voicemail Password

This is the password used to access the voicemail system (*98). It
can be changed by the user when they log into their voicemail (after
logging in, they dial 0 then 5).

Email Address

The address that voicemails notifications will be sent to

Pager email address

This email address will be sent a small message notifying of voicemail messages, suitable for an email-to-pager service.

Play CID

Read back caller's telephone number prior to playing the incoming
message, and just after announcing the date and time the message was
left.

Play Envelope

Envelope controls whether or not the voicemail system will play
the message envelope (date/time) before playing the voicemail message.
This settng does not affect the operation of the envelope option in the
advanced voicemail menu.

Delete Vmail

If set to "yes" the message will be deleted from the voicemailbox
(after having been emailed). Provides functionality that allows a user
to receive their voicemail via email alone, rather than having the
voicemail able to be retrieved from the Webinterface or the Extension
handset.

CAUTION: You must have email attachment set to yes if you
don't want your voicemail system to email you a notification saying
'You have a voicemail' and then immediately delete the voicemail. Make sure you've fully tested voicemail-to-email before you turn this on. See Email Problems for hints.

Feature Codes

Note: The following are from Asterisk@Home 2.7

extensions.conf

*411 — Directory

*78 — Do Not Disturb activate

*79 — Do Not Disturb deactivate

*98 — Check voicemail (enter extension)

*98nnn — Check voicemail (extension nnn)

*97 — Check voicemail (for calling extension)

*70 — Call Waiting activate

*71 — Call Waiting deactivate

*72 — Call Forward activate

*73 — Call Forward deactivate

*90 — Call Forward on Busy activate

*91 — Call Forward on Busy deactivate

*69 — Call trace

*11 — Log in (not used for fixed devices)

*12 — Log out (not used for fixed devices)

888 — Barge-In (unconfirmed, please update wiki)

8nnn — Meet-Me conferencing (nnn = 1 to 4-digit extension number)

*77 — Record announcement

*99 — Playback announcement

7777 — Simulate incoming voice call

666 — Simulate incoming fax call (unconfirmed, please update wiki)

*43 — Echo test

extensions_custom.conf

*60 — Time

*61 — Weather

*62 — Wakeup call

*65 — Playback extension

300 3nn 91npanxxxxxx — Set speed dial nn to 9-1-NPA-NXX-XXXX

3nn — Dial speed dial entry nn

*3nn — Listen to / dial speed dial entry nn

features.conf

#70 — Call Parking (Will park call on ext#71-79 and announce)

Follow Me

Follow-Me settings are just like a mini Ring Group, but it's tied directly to your extension. It's configured exactly the same way as a ring group, including the ability to use an announcement to alert people that they're being transferred elsewhere.

The dialplan has been structured such that any place where one might dial to ring an extension, the Follow-Me ringgroup will be engaged if it is defined. This means that from an IVR calling from outside and internally when dialing an extension you will reach the ring group in place of the extension. This behavior is enabled by simpling including the ext-findmefollow context ahead of the ext-local context any place that extensions are dialed.

There are other 'creative' uses of the Follow-Me function. At the simplest, you can simply put in the extension of the Follow Me number with a choice to go to its voicemail if not answered and you will be accomplishing exactly the same thing as if the extension was being dialed. However, you can now diverge with such simple things as changing your ring time to override the default, adding an announcement, going to an alternative voicemail or other destination if not reached, and of course adding multiple numbers and ring strategies when someone tries to call that number.

FreePBX 2.3 Screen Shots

This page contains Screen Shots from many common FreePBX Version 2.3 Module Screens

Adding Extensions

Digital Receptionist

Edit Administrator

Edit Recording

FreePBX System Status Dashboard

IVR Editing

Miscellaneous Applications

System Recordings

FreePBX Framework Module

The FreePBX Framework module is not like other modules on the system. Its purpose is to provide a mechanism to upgrade the overall Framework of FreePBX as described below.

Simple FreePBX Architecture Description

FreePBX is a modular architecture that can easily be extended with new modules to add functionality to the main system or extend existing functionality. In the absence of any module being loaded, there is still a foundation of code that is required to enable the system. This code base is referred to as the FreePBX Framework. Once the Framework is loaded, the system requires a minimal set of modules to provide any useful functionality. These include Core as well as a handful of other Key modules without which the system will not properly function. When FreePBX is initially installed, these modules are all installed at that time.

Module Administration in Framework

Part of the Framework code is the Module Administration Page. Once you have a Framework loaded, Adding modules is fairly straight forward. Under the covers, FreePBX simply explodes a module's tarball into the modules directory and then executes any included install scripts that the module requires.

Framework as a Module

In order to provide a mechanism that upgrades the Framework code, it was previously necessary to manually download a new version's tarball and then run install_amp which would upgrade you to that new version. Running install_amp is effectively takes files form the tarball and copies them onto the framework code in a similar manner to installing a module. (It does a bit more, but that is out of the scope of this description). The Framework module does effectively the same thing as install_amp. After being loaded by Module Admin it has a directory structure that is very similar to what you would see when untarring a FreePBX tarball. The Module Administration code then executes Framework's install script as if it were just another module. (As far as it know, it is just another Module). The install code proceeds to do almost the same thing that install_amp does resulting in all the Framework files being updated, and any incremental upgrade scripts being run. (It shares the same libraries as install_amp so it is basically the same code running.

FreePBX System Status

No documentation availabe. You can help by contributing to this book pate.

FreePBX Users & Devices

Description
FreePBX phone extensions are actually made up of two components which are not visible when you are running in the FreePBX default Extensions mode. However, each extension is made up of one FreePBX User and then zero or more devices. The number that you dial the user at is always the same and will typically be referred to as their extension number. That extension number can then be permanently associated with a fixed device in addition to being able to login to an adhoc device. Once logged into a device, it becomes that user's extension until logged off. It will use their CID and Outbound CID, DID's associated with that user will be routed to that device (and any others they are associated with), Voicemail MWI will display on that device, etc.
Details
THIS PAGE NEEDS TO BE WRITTEN, IT SHOULD COVER BOTH USERS AND DEVICES MENU ITEMS

Gabcast

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General Settings

Dialling Options

Aterisk Dial Command Options: See below.
Asterisk Outbound Dial Command Options: See below.

Dial command options

The most common options are 'tr', which means 'The person
receiving the call can transfer a call using #' and 'Generate ringtones
when an extension is ringing'. Other useful options are (Note, this
list is very incomplete. The complete list of options is huge and can
be seen by typing 'show application Dial' in the asterisk console):

• A(x) -Play an announcement to the called party, using 'x' as the file.
• D([called][:calling]) - Send the specified DTMF strings
  *after* the called party has answered, but before the call gets
  bridged. The 'called' DTMF string is sent to the called party, and the
  'calling' DTMF string is sent to the calling party. Both parameters can
  be used alone.
• h - Allow the called party to hang up by sending the '*' DTMF digit.
• H -Allow the calling party to hang up by hitting the '*' DTMF digit.
• r - Indicate ringing to the calling party. Pass no audio to the calling party until the called channel has answered.
• t - Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf.
• T - Allow the calling party to transfer the called party by sending the DTMF sequence defined in features.conf.
• w - Allow the called party to enable recording of the call by
  sending the DTMF sequence defined for one-touch recording in
  features.conf.
• W - Allow the calling party to enable recording of the call
  by sending the DTMF sequence defined for one-touch recording in
  features.conf.

Voicemail

Number of seconds to ring phone before sending callers to voicemail

This is reasonably self explanitory, except this number is also
used in Ring Groups to determine the amount of time each phone will
ring in a 'hunt' or 'memoryhunt' ring group. Setting 'Ring Time' in the
Extensions Module will have precedence over this setting. Generally there are
5 seconds per ring, so 20 seconds would be 4 rings.

Extension prefix for dialing direct to voicemail


Dialing this before an extension will not ring the extension, but send you directly to this person's voicemail box.

Company Directory

The company directory is reached by dialling '*411', or a user pushing '#' (if enabled) whilst in an IVR.

Find users in the Company Directory


When searching the Directory, this lets you chose wether it's searched by First name or Last name.


Play extension number


Plays a message "Please hold while I transfer you to extension
xxx" that lets the caller know what extension to use in the future. to
caller before transferring call. Useful if you have an indial range, or
wish to let users know which extension they can dial directory from the
IVR.

Operator Extension:

This is the number the callers (both internal users and callers calling in from the outside) will dial to get to the operator.

Inbound Routes

Information

The 'Inbound Routes' page lets you configure which destination
FreePBX uses for calls coming from Trunks. When a call is recieved by
Asterisk from a trunk, the DID and/or Caller ID is matched and the call
is dispached as per your settings.

DID Number

For a SIP or IAX peer, this is usually your Account Number. If you
have an account of '888123123', putting that in here will match calls
coming from that provider. Leaving this blank will match 'any'.

CID Number

The Caller ID number sent to your machine. This is not something
you should trust, as it is easily spoofable (both with Voice over IP
and normal telephone lines). Leaving it blank will, again, match any.

You can leave both of these blank to match any call, from any caller.

Fax Handling

With these two options, you can manage the way faxes are received
over this trunk. Note that VoIP and Faxing does not work well together,
and you most probably will have problems.

Privacy Manager

Turn this on to ask for the callers Caller ID if not provided.
This is useful for telemarketers, as they are loathe to divulge this
information and will usually hang up.

Options

Immediate Answer

This picks up the phone as soon as it rings (With Zaptel lines,
this happes after the Caller-ID is received, which may be up to three
rings). It then generates any further 'ring' tones, if required, down
the audio channel. Note that if you're using G729 or GSM, the rings
will sound funny to the caller.

Pause After Answer

The number of seconds we should wait after performing an Immediate
Answer. The primary purpose of this is to pause and listen for a fax
tone before allowing the call to proceed.:

Alert Info

ALERT_INFO can be used for distinctive ring with certain SIP
devices. The standard names are 'Bellcore-dr1' to 'Bellcore-dr7', Snom
phones can additionaly use a http:// url of a WAV or MP3 file.

Set Destination

This is a standard destination option group.

Inventory Database

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IVR (Digital Receptionist)

Information

The 'Digital Receptionist' page is the interface used to setup
your auto attendant when people call your business or home. Normally
heard as "Thanks you for calling MYBUSINESS, for Sales press 1, for
Service press 2", etc.

Getting Started

• For Pre-freePBX 2.1 IVR's, see (history)

When you select Digital Receptionst, the first page is now a
brief set of instructions on how to drive the IVR. You can either edit
an IVR, if one is existing, or create a new one by clicking on 'Add
IVR'.

Naming your IVR

Unlike the old Digital Receptionist system, this creates the IVR
(and calls it 'Unnamed') as soon as you click 'Add' - You'll see it
appear on the right straight away.

Your options are reasonable self-explanitary:

• Change Name: This is simply the descriptive name that appears on the right, and in the drop-down menu of Destinations
• Timeout: This is the amount of time the system waits before sending the call to the 't' destination
• Enable Directory: If you switch this on, users will be able to dial the FeatureCodes" title="Multiple pages with this name">feature code for Directory from IVR and access the Directory service.
• Enable Direct Dial: If you enable that, users will, in addition to being able to dial the IVR options, be able to directly dial an Extension number
• Announcement: A System Recording that is played to users when they enter the IVR. This can be set to 'nothing'

Configuring your IVR

In the box on the left, enter the option for the user. This may be
one, or a series of numbers, or, 'i', or 't'. 'i' and 't' have special
meanings:

• i: This is the destination used when a
  caller enters an invalid option - if you only have 1 2 and 3 defined,
  and they push 4, it will jump to this destination. The default option
  for this, ff you don't supply an 'i' destination, its to replay the
  current menu. If they hit 'i' more than three times, the call is hung
  up.
• t: This is the destination used when
  nothing happens. You might wish to have this one go directly to an
  operator, in case the caller doesn't have a DTMF phone. As with 'i',
  the default is to replay, and if it's been replayed three times, hang
  up.

Note that with freePBX 2.1, Destinations are only displayed if there is at least one entry in there. So if, for example, you have the DISA module enabled, but no DISA entries, it will not appear in the list.

The rest of the page should be self explanitory. Use 'Increase
Options' or 'Decrease Options' to alter the number of options
available. This won't let you decrease it to less than the number of
options that are currently set.

To delete an option, simply leave the selection blank.

When you're finished, click 'Save' and you have your new IVR.

Java SSH Terminal

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Miscellaneous Destinations

Overview

Misc Destinations allow you to use anything you could dial from a standard extension as a destination.

Example

You might want to have an IVR option that is 'If you want to speak
to rob, you can connect to his mobile by pushing 2', and having a Misc
Destination of

• Rob's Mobile
• 00402077155 (Note the leading 0, as that's what I use for an 'external' call)

Then in the IVR menu, you simply select 'Rob's Mobile' as a destination, and it will connect the caller through.

Music On Hold

Information

Here you can configure the Music On Hold files that will be
played. You can configure various 'Classes' of Music on Hold, which are
used in Queues. The idea behind that is your 'default' MOH is standard
music, and your various queues can have different 'hold' music while
they're waiting.

Uploading a file

Simply select 'browse' and pick a MP3 file on your system. Then click 'Upload'. It will appear in the list of MOH files below.

Online Support

Online support offers you two links, one to this Wiki, the other to a built-in IRC client

IRC Online Support

Clicking on this loads a Java IRC client, which connects to the
FreeNode? IRC network, and joins the '#freepbx' channel, where most of
the developers are.

NOTE

When you join the IRC channel, the FreePBX version you're using,
and the kernel version of your machine will be sent to everyone in the
channel automatically. This is to assist developers with diagnosing any
problems you have, but also may cause you to have privacy concerns. If
you do not wish this information to become public, DO NOT USE THIS
CLIENT.

Outbound Routes

Information

Outgoing calls are sent over trunks as determined by the
configuration of the Outbound Routing page. This is designed to be as
flexible as possible, and allows for fall-through and multiple paths -
eg, Least Cost Routing!

Adding a Route

Route Name

This is simply a descriptive name for the trunk, which will be shown on the right hand side of the screen.

Route Password

If this route is hit by a caller, and this is not empty, they will
be prompted for a password. If they get the password incorrect, the
call will be dropped, and will not try for a match on any further
trunks.

Emergency Dialling

Settnig this means that this route is used for 'Emergency' calls.
If you wish to have a different caller ID send for this call (eg, when
you're dialling 000/911/999), turn this on. Any calls matching this
dial pattern will use the Caller ID specified in Emergency CID rather
than the usual Outbound CID in Extensions.

Dial Patterns

A Dial Pattern is a unique set of digits that will select this trunk. Enter one dial pattern per line.

Rules:

• X - matches any digit from 0-9
• Z - matches any digit form 1-9
• N -matches any digit from 2-9
• [1237-9] - matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9)
• . - wildcard, matches one or more characters
• | - seperates a dialing prefix from the number (for example,
  9|NXXXXXX would match when some dialed "95551234" but would only pass
  "5551234" to the trunks)

Examples

• 000
  • Only use this route if the user has dialled '000' exactly.
• 9|911
  • Only use this route if the user has dialled '9911', but take off the 9 before sending it to the trunk
• 0|.
  • Any number that starts with 0, use this route

Trunk Sequence

When this route is matched by the Dial pattern above, trunks are
tried in the order listed here. Note that if you have a password
protected trunk, and the caller gets the password wrong, it does not
proceed to the next trunk. Make sure you click 'Add' after adding the
trunk, and before you click 'Submit'.

Paging and Intercom

Paging to Grandstream no longer works properly

Somone modified the page function, and the GXP-2000 Phones do not work
properly anymore while paging. I have provided a temp fix untill this
is sorted out.

If you add the following in extensions_custom.conf, then when you
dial 7243 or "PAGE" on your phone you will get a 2 way page on the SIP
extensions as entered below.

(You will need to change the SIP extensions entered below, to the ones you want to use for paging. )

If you want one way paging to many phones dial 7241 or "PAG1".

For simplicity, I assigned one of the programable buttons on all of my Grandstream GXP-2000's to dial 7243.

from-internal-custom

exten => 7243,1,SIPAddHeader(Call-Info: answer-after=0)

exten => 7243,2,Page(SIP/6331&SIP/5481&SIP/3361&SIP/6271&SIP/2741&SIP/7461&SIP/8431|d)

exten => 7243,3, Hangup

exten => 7241,1,SIPAddHeader(Call-Info: answer-after=0)

exten => 7241,2,Page(SIP/6331&SIP/5481&SIP/3361&SIP/6271&SIP/2741&SIP/7461&SIP/8431|)

exten => 7241,3, Hangup

Parking Lot

This module allows you to configure all the normal features.conf settings for the parking lot functionality of Asterisk.

These include:

• Enable/Disable the feature (while retaining settings)
• Parking Lot Extension
• Number of Parking Lot slots
• Parking Timeout before the call is returned to the orignal parker if not picked up
• Parking Lot Context (for advanced use)

The more useful part of this module is to specify a destination for
parked calls that get orphaned. This can occur if the call is not
picked up and for some reason the original parker can not be reached.
(e.g. the original parker is on the phone and does not have call
waiting or ignores it). In this case, call is diverted to the chosen
destination which is any of the standard destinations provided in all
modules that include such an option. Prior to sending the call to that
destination, you can configure the following options to further
identify the orphaned all:

• Parking Alert-Info (to provide a unique ring for the returned call)
• CallerID Prepend (to identify the call with additional CID information)
• Announcement (to be played to the orphaned caller to reassure them that you are trying to get them back to someone)

Parked Calls with BLF indicator light. -teknoprep- (you know you love me)

i have got this to work on my GXP-2000 phones using trixbox 1.1.1
on vmware with 1.2.12.1 compiled asterisk. so if you follow these
instructions you should be good to go. This tid bit of bash commands
was taken from X-Rob's lesson on
http://www.freepbx.org/2006/09/28/un-trixbox-your-trixbox/ .

• important.. IF YOU RUN VMWARE... do not do the first 2
  lines... NEVER run yum on your vmware box for updates if you run a
  trixbox vmware image. CentOS 4.4 does not cooperate well with vmware.

Now, lets get BLF working for Parked Calls.

cd /usr/src

wget http://aussievoip.com/storage/users/315/315/images/179/metermaid-1.2.7.1...

cd asterisk-1.2.12.1

patch -p0 < /usr/src/metermaid-1.2.7.1.txt

make

make install

asterisk -rx "stop now"

/etc/init.d/zaptel restart

amportal start

Now lets edit some files

nano /etc/asterisk/extension_custom.conf

add this to the file

exten => _*3, 1, ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|sip_incoming,${DIALEDPEERNUMBER},1)

;the line above this, NEEDS TO BE ON ONE LINE... not on 2

exten => 701,1,ParkedCall(701)

exten => 701,hint,Local/701@parkedcalls

exten => 702,1,ParkedCall(702)

exten => 702,hint,Local/702@parkedcalls

exten => 703,1,ParkedCall(703)

exten => 703,hint,Local/703@parkedcalls

exten => 704,1,ParkedCall(704)

exten => 704,hint,Local/704@parkedcalls

Save the file and now lets edit another file

nano /etc/asterisk/features.conf

now it should look something like this when you are done in the General context

parkext => 700 ; What ext. to dial to park

parkpos => 701-704 ; What extensions to park calls on

context => parkedcalls ; Which context parked calls are in

parkingtime => 5000 ; Number of seconds a call can be parked for (default is 45 seconds)

• the important part is for the parkext to = 700 and the
  parkpos to be 701 and up.. usually i would only need 4 parking lots for
  now since most phones just don't have enough buttons to support more...
  but hey you can do whatever you need.

now lets use it

when a call comes in you press Transfer - *3... this will park the
call... the asterisk box will call you back and tell you where its
parked... First call parked always goes to 701.

now on your phone setup one of your button light indicators for

Asterisk BLF 701

Asterisk BLF 702

Asterisk BLF 703

Asterisk BLF 704

(this is how i did it on my gxp-2000)

now when a call gets parked anyone with this setup will see the
light indicator light up for any of the 4 parking lots... this will
give you the ability to have the call picked up from anywhere on the
system that has this setup... just press the button.

• Problems - when i first boot up my GXP-2000 the lights for
  the BLF of Parked Calls will light up even tho no calls are parked...
  this will change once a parked call is picked up on that lot. (2) If
  you restart asterisk in any way... it seems that parked BLF stops
  working... ANYONE have a fix for that?

Phone Book and Phone Book Directory

No documentation available. You can help by contributing to this book pate.

PHP AGI Library Configuration

No documentation available. You can help by contributing to this book pate.

PHP Configuration Information

No documentation available. You can help by contributing to this book page.

PIN Sets

PIN Sets are a module that allow you to use a range of PIN's, rather than just one. This is currently only used by Trunks, but it potentially could be used in DISA or anything else that uses PIN's for authentication.

Print Extensions

This module will print a list of users and extensions. Handy for creating a company directory listing quickly.

Queue Priorities

Queue Priorities
This page needs writing, please contact us if you would like to write this page.

Queues

Information

Queues allow you to manage a large number of incoming calls, as
you would expect to have in a Call Center. This is very intelligent
application, and as such, it has a lot of configuration options.

Queue Setup

Queue Number

This is the number that can be dialled from any extension to be
put into the queue. This is also the same number you use when selecting
a destination. Agents (eg, the people receiving the call) log in and
out of the queue by dialling the number then a single asterisk to log
in, or two asterisks to log out (eg, 700* to log in, 700** to log out)

Queue Name

A short name for the queue. This is only used in the web interface for ease of identification.

Queue Password

If you are concerned about security, you can put a password on the
queue to stop just anyone from logging into it. After the Agent tries
to log in, he or she will be prompted for the password here.

CID Name Prefix

As an agent may be logged into more than one queue, it can be
useful to have a prefix on the Caller ID seen on the agent's phone, so
he or she knows which queue the call is coming from - eg, 'Sales:' or
'Tech:.

Static Agents

These are devices that are always logged into the queue. This is
useful if you have an agent that is not directly connected to the
FreePBX PBX, but is telecommuting. You can put their number in as it
would be dialled from an internal extension. (Note that his has changed from AMP
— You used to need to prefix the number with a #. This is no longer
needed). The number will be routed as if it was dialled from a normal
extension, so dial rules in Outbound Routing and trunks are matched as
per normal.

Queue Options

Agent Announcement

This is an announcement that is played to the Agent prior to
connecting in the caller. An example could be: "the Following call is
from the Sales Queue" or "This call is from the Technical Support
Queue". This is useful when agents don't have Caller-ID on their phone,
or don't look at it for the CID Name Prefix. These recordings are
managed by System Recordings.

Hold Music Category

This is the category of Music (or Commercial) played to the caller
while they wait in line for an available agent. Categories are set up
in On Hold Music

Max Wait Time

The maximum number of seconds a caller can wait in a queue before
being pulled out and set to the destination beow. Set to 0 for
unlimited, but that's not recommended.

Max Callers

The maximum number of people permitted to wait in the queue. If
this number is reached, any further people will be sent straight to the
destination below.

Join Empty

If you wish to allow callers to join queues that currently have no agents, set this to yes. This is not recommended.

Leave When Empty

If you wish to remove callers from the queue if there are no
agents present, set this to yes. If you have agents logging in and out
all the time, you may wish to set this to 'no', otherwise a good idea
is to set this to yes - everyone's gone home, and didn't get around to
answering the customer.

Ring Strategy

There are 6 ring patterns to chose from:

• ringall: ring all available agents until one answers (default)
• roundrobin: take turns ringing each available agent
• leastrecent: ring agent which was least recently called by this queue
• fewestcalls: ring the agent with fewest completed calls from this queue
• random: ring random agent
• rrmemory: round robin with memory, remember where we left off last ring pass

Agent Timeout

The number of seconds an agents phone can ring before we consider it a timeout.

Retry

The number of seconds we wait before trying all the phones again

Wrap-up-time

After a successful call, how many seconds to wait before sending a
potentially free agent another call. The default is 0, or no delay.
You'll probably have grumpy agents with that. Try setting it to 30
seconds.

Call Recording

Incoming calls to agents are recorded. They are saved to /var/spool/asterisk/monitor.

Caller Announcements

Frequency

How often to announce queue position, estimated holdtime, and/or
voice menu to the caller. Set to 0 to Disable Announcements totally.

Announce Position

Set to 'Yes' to announce position of caller in the queue.

Announce Hold Time

Should we include estimated hold time in position announcements?
Either yes, no, or only once; hold time will not be announced if it's
estimated to be less than 1 minute.

Voice Menu

After announcing Position and/or Hold Time, you can optionally
present an existing Digital Receptionist Voice Menu - eg If you'd like
to leave your name and number for a call back, please push * now. This
will not effect your position in the queue. This voicemenu must only
contain single-digit 'dialed options'.

Join Announcement

The announcement played to callers once prior to joining the queue. These recordings are managed by System Recordings.

Fail Over Destination

This is a standard destination that is used in an overflow/timeout condition, which is configured above.

Ring Groups

Information

This defines a 'virtual' extension that rings a group of phones
simultaneously, stopping when any one of them is picked up. This is
basically just a dumber version of Queues for those that don't need the
extra functionality of it.

Add Ring Group

Group Number

This is the number that is dialled from any extension that will make all of the phones in the group ring.

Ring Strategy

• ringall: ring all available channels until one answers (this is the default)
• hunt: take turns ringing each available extension
• memoryhunt: ring first extension in the
  list, then ring the 1st and 2nd extension, then ring 1st 2nd and 3rd
  extension in the list.... etc.

Extension List

List extensions to ring, one per line. You can include an
extension on a remote system, or an external number by suffixing a
number with a hash (#). ex: 2448089# would dial 2448089 on the
appropriate trunk (see Outbound Routes)

CID Name prefix

You can optionally prefix the Caller ID name when ringing
extensions in this group. ie: If you prefix with "Sales:", a call from
John Doe would display as "Sales:John Doe" on the extensions that ring.
(Note, you can't use a space here.)

Ring Time

How long (in seconds) the group of phones will ring before
'failing' and doing the options specified below. This is not related to
the 'hunt' ring strategy above, but is the total length of time a call
will stay in the group before using the 'Destination if no answer'
selection.

Destination if no answer

This gives you a selection of things to do when the call exceeds
the 'Ring Time' specified above. Options may be added and removed from
this list depending on the modules that are installed. See Destinations for more information.

System Recordings

System Recordings are used in Ring Groups and Conferences for various announcements.

Uploading a file

If you're uploading a .wav file directly, it needs to be saved as a 'PCM Uncompressed' 8000hz 16bit mono file.

Time Conditions

Information

Time Conditions are a module that appears as a destination when
installed. It allows you to do an 'if' based on the current Time,
Weekday, Day of the Month, or Month. At the moment it's reasonably
basic with no support for 'AND' or 'OR', but you can chain together
time conditions to do the same thing.

Time Condition Name

A short name which is used to identify the Time Condition in a destination.

Time Condition

Select from the pull down menus the time range that you want to use. Note that '-' means 'Any'.

Destinations

You have a choice of two destinations,
depending on wether the time is matched or not. You can chain together
time conditions to get an 'OR' or 'AND' effect. Unfortunately, there's
no easy way to do that... at the moment!

Example of OR

Lets say you want Monday to Friday, 9am to 5pm or Saturday 9am to 12pm to go to IVR1, whilst all the rest of the time, it goes to IVR2.

• Timecond1 Match Mon-Fri, 9am-5pm
  • True: Goto IVR1
  • False: Goto Timecond2
• Timecond2: Match Sat, 9am-12pm
  • True: Goto IVR1
  • False: Goto IVR2

Example of AND

On Between 9am and 5pm on Monday and Wednesday, go to IVR2. Otherwise go to IVR1.

• Timecond1: Match 9am-5pm
  • True: Goto Timecond2
  • False: Goto IVR1
• Timecond2: Match Monday
  • True: Goto IVR2
  • False: Goto Timecond3
• Timecond3: Match Wednesday
  • True: Goto IVR2
  • False: Goto IVR1

Time Groups

Time Groups
Needs Content, please contact us if you would like to help write this page.

Trunks

Upgrade Module

Upgrade Module Overview

The Upgrade Module is a module that is used to allow FreePBX to be upgraded from one major version number to a higher version. For example, to upgrade from version 2.3.X to 2.4.0. The way that FreePBX manages the Online repository is by providing an XML file associated with each X.X version of the product. FreePBX uses this XML file to detemine what the latest available modules are for your current version and where to find them. It then compares this against what you have installed to allow you to upgrade.

In order to upgrade from a version such as from 2.3.X to 2.4.X you need to have Module Admin download modules-2.4.xml instead of modules 2.3.xml. Since FreePBX uses its internal version number to detemine which one to pull, the upgrade process simply bumps the current FreePBX version to the lowest possible version so that it will download the proper XML file. The lowest version, in the case of 2.4, would be 2.4.0alpha0 which in fact never existed since we always start with version 1 for alpha, beta and RC releases.

Once you have bumped the version number, you need to go through the process of downloading modules. The Framework module is a special module that often has libraries and functions required by many other modules. So this module usually needs to be downloaded first, followed by all the other modules. Also - it is important during the upgrade process that you don't press the Apply Configuration bar. The Upgrade module has a mechanism that should keep the Apply Configuration bar from coming up until you have finished the upgrade process. This is to make sure you keep from running the configuration generation part of FreePBX (retrieve_conf) which may have other requirements not yet upgraded until all modules are brought up to date.

Upgrade Process

During each step of the upgrade process the top display of the Upgrade module will show the current version for FreePBX Base, FreePBX Framework and FreePBX Core since these are typcially the core critical modules that need to get fully upgraded before the module is satisified and removes itself. You should still be certain to upgrade ALL other modules that have upgrades available or you will have potential for errors.

The step that are involved in the upgrade process are:

1. One might acquire the 2.4 Upgrade Tool through many methods but most will simply get it through the familiar Module Admin by clicking on the “Check for updates online” link/button.
2. Install the module called "2.4 Upgrade Tool," by clicking on the module and selecting Install. Then click the process button at the bottom of this screen.
3. Now we will see a typical module install screen requesting that we confirm the “2.4 Upgrade Tool 2.3.0 will be installed and enabled.” Click the “Confirm” button now to proceed.
4. Once this tool is installed a new option will show up on the left navigation bar under the Admin Section titled “2.4 Upgrade Tool.” Think of this temporary tool as your guide through a healthy install process. Only once the upgrade process is complete and no longer necessary will this tool handily remove itself from the navigation bar.
   
   

5. We will return to this button several times as it will guide us through the complete multistep process dynamically providing status updates and instructions as we proceed. Go ahead and click the “2.4 Upgrade Tool” button now.
6. Please take the time to read through the provided instructions.
   
   
7. Please note, as tempting as it may be, the repeated instructions of IMPORTANT: Do NOT Apply Configuration Changes (reload bar) until you have gone through all the steps!
8. Click on the Upgrade Now button and confirm the first part of the upgrade process at this time.
9. Read the instructions (RTMF) as they have dynamically updated. Notice that the line FreePBX Base Version: now has a new value of “2.4.0alpha0.” Each of these three components, Base, Framework, and Core will display their currently respective versions on the local system and be updated as we progress.
   
   
10. The instructions now say to go back to Module Admin and choose to ONLY upgrade “FreePBX Framework.” Do so now following the usual upgrade confirmation and return prompts.
    
11. Now we may upgrade Core and everything else in this step. Once again do so now.
12. As previously stated we have repeated been told to NOT press the reload bar until you have completed the previous step. We may do so now.
13. Upgrade Complete.
14. Notice that the “2.4 Upgrade Tool” has removed itself from the navigation bar as it has completed its task of guiding us through the easiest FreePBX upgrade to date.

Once you have completed these step you will be upgraded to the new version. Under normal circumstance, the module should delete itself. If it is still there, try clicking on it and see if that removes it. If not, read the status information to see if you have really finished the upgrade. If you have it should go away.

VoiceMail Blasting (VMB)

VoiceMail Blasting VMB

In Progress

Voicemail Module

The Voicemail Module is a required for the system to work properly. For
that reason, once installed it can not be un-installed. This module
provides the functionality to the Extensions or Users Page to configure
Voicemail. It also provides the Feature Codes used to dial into a users
Voicemail box.

ZAP Channel DIDs

ZAP Channel DIDs

In Progress

ZOIP

ZoIP - The Great Asterisk Underground Empire

Originally from http://uc.org/read/ZoIP

Zoip is a new way of playing Zork - It uses Text-to-Speech and
Speech Recognition to play the classic infocom game, Zork. It appears
as a destination. If you wish to play this without needing to go through an IVR,
the easy way is to create a Ring Group with an invalid extension in it, and a destination of Zoip.

Also, if your FreePBX machine is on a slow connection, BE WARNED. This is a 7 Megabyte
download. That's a lot through a slow connection. You don't need to use
the 'online modules' link to donwload this, you can download it
manually - see the bottom of this page, with the title of 'Downloading
it Manually'. I can't download it through a 1.5mbit connection in under
the default 30 second max execution time. If you don't want to download
it manually, and the screen just goes white and stops after 30 seconds,
this is because your 'max_execution_time' in /etc/php.ini is too short.
Set it to 600 (10 minutes), restart apache (/etc/init.d/httpd restart as root) and it should work fine.

As it does some new, non-standard things, there are some packages
that need to be installed. Without these, the module will not work
properly.

CenOS/RHEL/FC3/4/5 Instructions

From the shell:

That's it - you can now download the module and it'll work happily!

Debian, Ubuntu, apt-style distros

This is taken from the Zoip installer, and may not be accurate. I don't have a Debian machine here, so I can't test this.

• Install Sphinx2, and Festival
  • Install the Sphinx2 binaries and developer kit

• • Install the Sphinx2 Perl bindings

• • Install Festival, and the 'kallpc8k' voice, sitable for telephone quality

• Add some additional perl modules used by ZoIP from the apt repository

Installing ZoIP Manually

Beacuse this is abig file, it may be neccesary to download the module manually. This is simply done by:

From here, you can go back to your Module Administration page, and
the Zoip module will appear as a 'Disabled Module' ready to be
installed, without requiring going to the online module repository.

Third-Party Unsupported Modules

Contributed Modules

These modules are unsupported — they have been known to cause happiness and glee, or dizziness, confusion, and frustration.

The modules can be obtained from:
http://mirror.freepbx.org/modules/release/contributed_modules/

Instructions for installing third-party modules that are not yet included in the Third-Party module repository (e.g. recently contributed modules) can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/modules/README.t...

Please understand that these modules were written by other users of FreePBX and not the FreePBX development team. Issues such as what version or versions they will work with, problems with installation, etc. need to be directed to the authors of these modules and not the FreePBX development team.

If you have installed any of these modules and are about to upgrade your system from one version to a newer version it is possible that right after the upgrade your system might not work properly. If that is the case please first disable these third party modules to verify that they are not causing the issue. Some authors don't upgrade versions as quickly as others do and might not know there is a problem.

It is also possible that when you upgrade that suddenly you might not have a working GUI to disable a module. If that becomes the case, you can do the following at a Linux prompt (assuming standard install defaults):

/var/www/html/admin/modules/framework/bin/module_admin disable {module name}

If you are an author of any module(s) here we request that you please keep it updated, and in your description include what version(s) it has been designed and tested on, so that in the future people will know how current the posted module is when they look at it.

(est),NoOp(Speaking Clock for Eastern Time Zone)
Set(TimeZn=EST5EDT)
goto(scstart)
(cst),NoOp(Speaking Clock for Central Time Zone)
Set(TimeZn=CST6CDT)
(scstart),Ringing
Set(FutureTime=$[${EPOCH} + 8])
Set(FutureTimeMod=$[${FutureTime} % 10])
Set(FutureTime=$[${FutureTime} - ${FutureTimeMod}])
Set(MaxConnectTime=$[${FutureTime} + 180])
(scringsomemore),Set(FutureTimeMod=$[${FutureTime} - ${EPOCH}])
GotoIf($["${FutureTimeMod}" < "0"]?scanswer:scwaitasec)
(scwaitasec),wait(1)
goto(scringsomemore)
(scanswer),Answer
(scplayagain),Set(FutureTime=$[${FutureTime} + 10])
Set(FutureTimeMod=$[${FutureTime} % 60])
wait(1)
playback(at-tone-time-exactly)
SayUnixTime(${FutureTime},${TimeZn},IM)
GotoIf($["${FutureTimeMod}" = "0"]?scexactmin:scsaysecs)
(scexactmin),SayUnixTime(${FutureTime},${TimeZn},p)
goto(scwaittobeep)
(scsaysecs),playback(and)
SayUnixTime(${FutureTime},${TimeZn},S)
playback(seconds)
(scwaittobeep),Set(FutureTimeMod=$[${FutureTime} - ${EPOCH}])
GotoIf($["${FutureTimeMod}" < "1"]?scplaybeep:scwaitsectobeep)
(scwaitsectobeep),wait(1)
goto(scwaittobeep)
(scplaybeep),playback(beep)
Set(FutureTimeMod=$[${MaxConnectTime} - ${EPOCH}])
GotoIf($["${FutureTimeMod}" < "1"]?scthatsall:scplayagain)
(scthatsall),GotoIf($["x${IVR_CONTEXT}" = "x"]?app-blackhole,hangup,1:${IVR_CONTEXT},return,1)