FreePBX is an Asterisk management system with a web interface.

See also: Other FreePBX pages in this wiki

The FreePBX administration console:

http://ip.of.your.pbx

If this is the first visit to the FreePBX web admin page, click “Apply Configuration Changes” and reboot the new PBX again.

Save each change and click Apply Configuration after done making changes.

FreePBX → Admin → Administrators → admin →

• Password → newfreepbxadminpassword

FreePBX → Settings → Advanced Settings → Asterisk Manager →

• Asterisk Manager Password → your-asterisk-manager-password

FreePBX → Settings → Advanced Settings → System Setup →

• User Portal Admin Username → newariadminusername
• User Portal Admin Password → newariadminpassword

FreePBX → Admin → Module Admin → Check Online → Upgrade All → Process

Verify that the username and password in /etc/asterisk/manager.conf and /etc/amportal.conf match.

http://www.freepbx.org/support/documentation/faq/changing-the-asterisk-manager-password

If you see an error during FreePBX installation like:

Checking for PEAR DB..FAILED

try:

pear install DB

then re-run:

./install_amp

If you enable remote access to your PBX, secure it!

NAT is a real hurdle for SIP. The best way to deal with NAT issues is to not use NAT if at all possible. NAT on both ends may not be worth attempting if using SIP, it's just not a NAT-friendly protocol like IAX.

Using phones with IAX protocol support is a good alternative if the PBX is behind NAT.

IAX protocol is pretty much Asterisk-specific.

If your PBX is behind NAT, forward the single UDP port 4569 from your NAT firewall in to the PBX.

http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension

If your PBX is behind NAT and you'd still like to try getting remote SIP extensions to work:

vim /etc/asterisk/sip_nat.conf

localnet=192.168.1.0/255.255.255.0      #your local network
externhost=your.fqdn.hostname           #your resolvable host name
fromdomain=your.fqdn.domain.name        #your domain mane
nat=yes
qualify=yes
externrefresh=10
canreinvite=no

asterisk -rx reload                     #reload Asterisk configuration

This should be the default:

vim -c 457 /etc/php.ini

memory_limit = 128M

amportal stop

rm -f /etc/asterisk/{sip_notify.conf,iax.conf,logger.conf,features.conf,sip.conf,extensions.conf,ccss.conf,chan_dahdi.conf}

/usr/sbin/safe_asterisk

cd /usr/src/freepbx-2.10.0
./install_amp

Visit the configuration page at: http://IP.of.PBX

Click “Apply Settings”

Reboot

http://www.freepbx.org/support/documentation/installation/first-steps-after-installation

FreePBX offers numerous add-on modules.

• You probably don't want or need to install them all
  • Simpler user interface
  • Enhanced security
• If you are looking for a feature and don't find it
  • FreePBX → Admin → Module Admin → Check Online

Commonly installed modules:

• Ring Groups
• IVR
• Backup and Restore
• Follow Me
• Asterisk Info
• Asterisk Logfiles
• Asterisk SIP Settings
• OSS Endpoint Manager

Install the Sip Settings FreePBX module, if it's not already installed, then:

Settings → Asterisk SIP Settings

If you have Postfix installed (default in CentOS 6), it's easy to use that:

See also Postfix Authenticated Smarthost

If you don't have an MTA installed, SSMTP is a simple alternative:

See also SSMTP

Install either the free OSS End Point Manager or the commercial (and more capable) End Point Manager FreePBX module.

http://www.the159.com/endpointman/tut.html

Add a regular SIP extension (phone):

Applications → Extensions → Add Extension → Generic SIP Device

http://www.freepbx.org/support/documentation/howtos/howto-route-dial-patterns-and-trunk-dial-rules

http://www.inphonex.com/support/trixbox-configuration-v2.6.1.1.php

Trunk Description: InPhonex Outbound Caller ID: 5201231234

Dial Rules only modify dial strings. Use '+' to add or '|' to remove digits:

1520+NXXXXXX 1+NXXNXXXXXX

Trunk Name: inphonex-outbound Peer Details:

type=peer insecure=very host=sip.inphonex.com username=yourinphonexusername secret=yourinphonexpassword qualify=yes sendrpid=yes context=from-pstn fromuser=yourinphonexusername fromdomain=sip.inphonex.com canreinvite=no

User Context: inphonex-inbound

User Details:

type=friend context=from-pstn username=yourinphonexusername user=yourinphonexusername insecure=very host=sip.inphonex.com fromdomain=sip.inphonex.com

Registration String:

yourinphonexusername:yourinphonexpassword@sip.inphonex.com/yourinphonexusername

Trunk Description: Vitelity Outbound Caller ID: 5201231234

Dial Rules only modify dial strings.

Prefix digits are stripped. Prepend digits are added to the dialed digits.

Vitelity requires 11 digits (1 + area code).

1520+NXXXXXX
1+NXXNXXXXXX

Trunk Name: vitelity-outbound
Peer Details: 

type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=yourvitelityusername
fromuser=yourvitelityusername
trustrpid=yes
sendrpid=yes
secret=yourvitelitypassword
allow=all
canreinvite=no

User Context: vitelity-inbound

User Details:

type=friend
dtmfmode=auto
host=inbound23.vitelity.net
context=inbound
username=yourvitelityusername
secret=yourvitelitypassword
allow=all
insecure=very
canreinvite=no

Registration String:

yourvitelityusername:yourvitelitypassword@inbound23.vitelity.net:5060

Route Name: Default Emergency: enabled Dial Patterns:

011. 1NXXNXXXXXX NXXNXXXXXX NXXXXXX

Pick a trunk or two.

Inbound Routes

Route Name: Default Set Destination: