====== FreePBX Distro ======

**Home**: http://www.freepbx.org/freepbx-distro

**Pro Tips Series Videos**: https://www.youtube.com/results?search_query=freepbx+open+source+pro+tips+sangoma

**Downloads**: https://downloads.freepbxdistro.org/ISO/

The FreePBX distro is:

  * A complete bare-metal iPBX
  * Based on CentOS 5.7
  * Includes Asterisk 1.8
  * Includes Fail2Ban
  * Initial memory usage is under 130MB
  * iSymphony is integrated as the Operator Panel

===== Updating =====

http://wiki.freepbx.org/display/PPS/Updating+FreePBX+Official+Distro

http://www.freepbx.org/support/documentation/installation/updating-the-freepbx-distro

===== Installation =====

Netinstall: http://www.freepbx.org/download-freepbx

http://wiki.freepbx.org/display/FD/Installing+FreePBX+Official+Distro

http://www.freepbx.org/support/documentation/installation/installing-the-freepbx-distro

  * Set the first interface to DHCP if possible
  * Set a DHCP reservation, or static IP addressing if that's not convenient
  * Set the root password during install
  * Set the administrative username and password on first login with a web browser

===== First Steps =====

http://wiki.freepbx.org/display/FD/FreePBX+Distro+First+Steps+After+Installation

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

==== RTC Clock ====

You may have problems shutting down or rebooting your PBX where the system hangs for over 5 minutes trying to shut down MySQL or MariaDB.

<file>
A stop job is running for MariaDB...
</file>

This change to the configuration will fix this:

<file>
timedatectl set-local-rtc 0
</file>

===== Trunks =====

==== DAHDI (ZAP) ====

FIXME

http://www.pbxinaflash.com/community/index.php?threads/install-aex800-wildcard-using-dahdi-on-piaf-purple.11214/

  * DAHDI is the newer interface for TDM cards
  * ZAP is the old name
  * When naming a ZAP/DAHDI trunk
    * 'g0' will occupy lines in ascending order
    * 'G0' will occupy lines in decending order

==== Mediatrix ====

FIXME

http://www.sonoracomm.com/support/20-voice-support/154-mediatrix-1204

http://lists.digium.com/pipermail/asterisk-users/2004-February/028568.html

	Create a trunk
		Mediatrix gateway is SIP trunk
			Name, Description, Host, Codecs
			No authentication
	Routes -> Inbound Routes
		Create a new inbound route
			Choose a DID number to use (such as 999)

Also, you can create a new Feature extension and use that.  Security implications?

	On the Mediatrix 1204 gateway
		No authentication
		Point ports to extension 999 on server for inbound calls
		See setup info on web site for SAIL
		Logging
			Device Info -> Monitoring
				Enable syslog to PBX IP
			On PBX, enable remote logging
				vi /etc/sysconfig/syslog
					add '-r' to SYSLOGD_OPTIONS
				service syslog restart

==== Vitelity ====

  * You can use Vitelity either as a registered or an IP-routed trunk
  * Vitelity recommends using the more efficient IP-routed method to connect PBXs with static IPs
  * You must specify a single CLID for a 'registered' trunk using the Vitelity web admin console
  * You can manipulate the CLID normally for 'IP-routed' trunks

=== PBX With Dynamic IP ===

^Caller ID^Number^Comment^
|Outbound CLID|XXXXXXXXXX|Extension CLID overrides trunk CLID|

^Prepend^^
|1520|NXXXXXX|
|1|NXXNXXXXXX|

^Outgoing Settings^
|vitelity-outbound|

<file>
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=your_vitelity_login_name
fromuser=your_vitelity_login_name
trustrpid=yes
sendrpid=yes
secret=your_vitelity_password
allow=all
nat=yes
canreinvite=no
qualify=yes
</file>

^Incoming Settings^
|vitelity-inbound|

<file>
type=friend
dtmfmode=auto
host=inbound23.vitelity.net
context=from-trunk
username=your_vitelity_login_name
secret=your_vitelity_password
allow=all
insecure=very
nat=yes
canreinvite=no
qualify=yes
</file>

^Registration^^
|Register String|your_vitelity_login_name:your_vitelity_password@inbound23.vitelity.net:5060|

=== PBX With Static IP ===

FIXME

IP-routed method shown here with separate trunks for inbound and outbound.

Note that you can only manipulate outbound caller ID if using IP routing.  If using registration, specify the desired CLID in the Vitelity Portal Subaccount.

	Create SIP Trunk
	Name: vitel-inbound
	Description: Vitelity Inbound SIP Trunk
	Host: inbound23.vitelity.net
	Direction: Inbound
	Context: from-outside
	Caller ID: Your Company <5551234567>
	Codecs: G.711u, G.729, GSM 			# order by preference
	DTMF mode: Auto
	Other options: insecure=very
	Provider's web site	https://portal.vitelity.net

Clone the inbound trunk to create the outbound trunk:

	Name: vitel-outbound
	Host: 64.2.142.93
	Description: Vitelity Outbound SIP Trunk
	Direction: Outbound
	Context: from-inside
	Codecs: G.711u, G.729, GSM
	DTMF mode: Auto
	Provider's web site	https://portal.vitelity.net

===== Day/Night Mode =====

http://www.freepbx.org/forum/freepbx/tips-and-tricks/blf-key-for-day-night-mode-or-call-flow-control

===== Xen Virtualization =====

:!: Unsupported!  The FreePBX distro is focused on a simple, reliable, 32-bit implementation.

  * No Xen kernel available
  * Runs as HVM
  * Won't run XenServer Tools