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Table of Contents
FreePBX Configuration
Incomplete
FreePBX is an Asterisk management system with a web interface.
See also: Other FreePBX pages in this wiki
Using FreePBX
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.
Default Credentials
| Function | Username | Password | Comment |
|---|---|---|---|
| FreePBX | admin | admin | |
| Voicemail & Recordings (ARI) | <none> | <none> | Use the FreePBX admin console to enable |
Configuration
Important Initial Settings
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
Security
Troubleshooting
Asterisk Manager Interface
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
Pear DB
If you see an error during FreePBX installation like:
Checking for PEAR DB..FAILED
try:
pear install DB
then re-run:
./install_amp
Remote Extensions
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.
IAX Protocol
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.
SIP Protocol
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
PHP Memory Limit
This should be the default:
vim -c 457 /etc/php.ini
memory_limit = 128M
Re-Install Just FreePBX
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
First Steps
Add-On Modules
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
NAT
Install the Sip Settings FreePBX module, if it's not already installed, then:
Settings → Asterisk SIP Settings
Send E-Mail
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
Phone Management
Install either the free OSS End Point Manager or the commercial (and more capable) End Point Manager FreePBX module.
Extensions
Add a regular SIP extension (phone):
Applications → Extensions → Add Extension → Generic SIP Device
Trunks
Vitelity
Add separate inbound and outbound routes.
Inbound
General Settings
| Trunk Name | vitel-inbound |
| Outbound Caller ID | Your Name <5201231234> |
Outgoing Settings
| Trunk Name | vitel-inbound |
Peer Details
type=friend dtmfmode=auto username=yourvitelityusername secret=yourvitelitypassword context=from-trunk insecure=port,invite canreinvite=no host=inbound23.vitelity.net
Registration String
yourvitelityusername:yourvitelitypassword@inbound23.vitelity.net:5060
Outbound
General Settings
| Trunk Name | vitel-outbound |
| Outbound Caller ID | Your Name <5201231234> |
| Maximum Channels | Set to the number of simultaneous calls you expect on a regular basis. |
Dialed Number Manipulation Rules
These 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
Outgoing Settings
Trunk Name: vitel-outbound
Peer Details:
type=friend dtmfmode=auto username=yourvitelityusername secret=yourvitelitypassword fromuser=yourvitelityusername trustrpid=yes sendrpid=yes canreinvite=no host=outbound.vitelity.net
Registration
If your PBX will use IP Routing (recommended by Vitelity), do not enter a registration string.
If your PBX is behind a dynamic IP address (which changes occasionally), you do need a registration string.
Registration String:
yourvitelityusername:yourvitelitypassword@inbound23.vitelity.net:5060
Routes
Inbound
Description: Default
DID Number: The called number; usually your account number
Set Destination: Usually an extension, ring group or IVR
Outbound
Route Name: Default
Emergency: Enabled if this route is used to call 911
Dial Patterns
InternationalLong distanceLong distance (no 1)LocalEmergency(only if e911 service is on this trunk)
+ |011.
+ |1NXXNXXXXXX
+ |NXXNXXXXXX
+ |NXXXXXX
+ |911
Trunk Sequence for Matched Routes
Pick a trunk or two (vitel-outbound).
