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====== Thirdlane iPBX ======

Thirdlane PBX is a professional GUI management tool for Asterisk.  There is a multi-tenant version available.

===== Searching the Forums =====

Searching with google may be preferable to the built-in forum search.  Preface a google search with:
<file>
site:http://www.thirdlane.com/forum
</file>

===== License =====

Licenses are keyed to a hardware MAC address.

If you install the PBX from the .iso, a 30-day license is created.

If you create a backup, then delete these files from the .tar.gz, the temporary license will not be overwritten:

<file>
/etc/webmin/asterisk/license-data
/etc/webmin/asterisk/pbxmanager.license
</file>

==== Spoof a MAC Address ====

You can spoof the hardware MAC address after a hardware change.

<file>
vim /etc/rc.local

ifconfig eth0 down hw ether 00:07:E9:DA:56:01	#specify desired MAC address
ifconfig eth0 up
</file>

===== Upgrade Thirdlane =====

From 'Home' page, click the 'check for updates link.

There is always a chance that something will break.

Don't forget to reload.

===== Upgrade Asterisk =====

Be very careful about upgrading the host OS or Asterisk.  Avoid doing so if possible.

==== Recompile Asterisk ====

This probably doesn't apply if you used the Thirdlane ISO to install the PBX.

For **asterisk** and **asterisk-addons** you should:

<file>
make distclean
./configure
make menuselect
make
make install
</file>

===== Tools =====

Enter 'pbx-help' at the command line for a list of some tools.

===== Installation =====

Thirdlane can be installed via an ISO (CD) that packages CentOS, Asterisk and various goodies such as vTiger CRM and the Thirdlane web management interface.  Or, you can install any Linux, Asterisk, Webmin plus the Thirdlane Webmin GUI module.

Thirdlane maintains a CentOS 5 repository so updates are easier than before (untested).

Configure a [[networking:linux:static_ip_addr|static IP address]] and default route.

Setup DNS to resolve host 'pbx' to the internal IP address.

<file>
yum update -y
</file>

==== MySQL ====

If you install via the Thirdlane ISO, the default MySQL password is "passw0rd".

Root password is empty unless you set it:

<file>
mysqladmin -u root password 'mypassword'
</file>

Reset lost mysql root password:

<file>
service mysqld stop

mysqld_safe --skip-grant-tables &

mysql -u root
  use mysql;
  update user set password=PASSWORD("new-password-here") where User='root';
  flush privileges;
  quit
  
service mysqld restart
</file>

==== DAHDI ====

No installation required when using the ISO install as the ISO uses RPM-based packages and Thirdlane repo.

=== Configuration ===

See **[[voice:pbx:dahdi|DAHDI]]**

===== NTP =====

Make PBX ntp server for phone network:

<file>
vim /etc/ntp.conf

# Hosts on local network are less restricted.
restrict 192.168.30.0 mask 255.255.255.0 nomodify notrap
broadcast 192.168.30.255

service ntpd restart

netstat -uapn|grep 123
</file>

===== DHCP Server =====

FIXME

===== Music on Hold =====

See **[[voice:pbx:moh|Music on Hold]]**

===== Configuration =====

==== System Settings ====

	Branding
		Upload Logo (150x60 px)
		Logo URL
	Network
		eth0 (LAN)
			Addressing
				192.168.1.30
			Hostname
				FQDN
			DNS Domain
		Postfix SMTP address rewriting
			iPBX@sonoracomm.com
				domain name of owner
				important for mail relaying
	Provisioning
		TFTP, HTTP and FTP
			all enabled by default
		This server's address - enter server IP address
			external server IP if it has one
	Preferences
		Color - Lilac
	License
		Upload the license file sent by Thirdlane

==== PBX Settings ====

	Default Values
		Caller ID
		Default Script
		Distinctive Ring
			 yes 
			 Internal (Polycom)
		Voicemail Template
			 Probably OK as-is
		Phone Template
			 
	Global Variables
		OPERATOR
		RECORDING_FORMAT (wav for non-MS)
		Schedules
			tl-off-hours
				Office Closed Hours
				00:00-07:59
				17:00-23:59
				holidays
==== Media Files ====

		Music on Hold
			Use 'default' or create a new playlist
			Default MoH is not random
			Delete songs you don't like
				Delete all but 'morning coffee' and/or 'simplicity'
			Add songs you do like to a new playlist

==== Trunks ====

=== DAHDI (Zap) ===


	Create a trunk
		Name your first trunk 'G0' (case sensitive)

=== Mediatrix ===

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.

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: Company Name <NXXNXXXXXX>
	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: <outbound_ip_address>
	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

==== Extensions ====

  * Bulk creation is easy
  * 468* for a few seconds to reset Polycom phones
    * Hit 'Setup'
      * Set server to TFTP
      * Set server IP address
  * Directory - may need tweaking

==== Routes ====

  * Add DIDs, if using digital circuit or VoIP trunks
  * Add channel mappings for analog circuits, if desired

=== Inbound ===

  * Just modify pattern 's' if using FXO gateway
    * Office Closed Hours -> Play Closed IVR
    * All Hours -> Play Open IVR, Except when OFFICE_OPEN_OVERRIDE
    * All Hours #2 (cloned from tl-allhours) -> Play Open IVR

You can create a new inbound route for an extension, but it is easier to just add the DID number in the User Extension instead of doing this.

  * DID
    * Create Schedule-based Route
    * DID/Pattern: <your_did_number>
    * Description: My Direct Dial
    * When: All Hours
    * Script: Goto User Extension
    * User Extension: 101

=== Outbound ===

  * Point existing routes to new trunk just created
  * Use check boxes and Assign button

==== PBX Features ====

http://www.thirdlane.com/forum/selective-call-recording-by-pressing-9

	Feature Codes
		_8NXXX	tl-transfer-to-voicemail, enter '1' and 'u', Forward to VM
		*96		Close Office During Normally Open Hours (operator), closeoffice
		*97		Reopens Office During Normally Open Hours (operator), openoffice

===== Tools =====

	Auto-Provisioning
		Configure or fine-tune phones and MAC addresses
		Hit 'Provision Selected' to actually create config files and reboot phones

===== IVR (Auto-Attendant) =====

FIXME
	
===== Backup =====

	Tools -> Backup
	Create user on SME Server
	Enable FTP on SME Server
	Schedule backup
		Simple -> Daily
		FTP Server -> <ftp_server_name>
		File on FTP Server -> /path/to/your/backup.thirdlane.tar.gz

I also had to edit root's crontab to suppress useless e-mail messages sent nightly:

<file>
crontab -e

@daily /etc/webmin/asterisk/backup.pl >/dev/null 2>&1
</file>

===== Restore =====

Asterisk CLI problem after restore:

Edit the path in asterisk.conf to say /var/run/asterisk and restart:

<file>
vim /etc/asterisk/asterisk.conf
pbxportal kill
pbxportal start
rm /var/run/asterisk.*
asterisk -vvvvvvvr
</file>

===== Security =====

==== Fail2Ban ====

**[[networking:linux:fail2ban|Fail2Ban]]** is installed by default using TL ISO 2.0+.

:!: Fail2ban can complicate troubleshooting, so check it!

==== Arno's Firewall ====

<file>
vim /etc/arno-iptables-firewall/firewall.conf

OPEN_TCP="2222"
OPEN_UDP="4569,5060-5061,10000-20000"
HOST_OPEN_TCP="209.193.93.158~80,443,10000 209.193.64.254~80,443,10000 70.162.22.135~80,443,10000 209.193.64.2~80,443,10000"

iptables -nL |grep ACCEPT

ACCEPT     tcp  --  0.0.0.0/0            0.0.0.0/0           tcp dpt:2222 
ACCEPT     udp  --  0.0.0.0/0            0.0.0.0/0           udp dpt:4569
ACCEPT     tcp  --  209.193.93.158       0.0.0.0/0           tcp dpt:80 
ACCEPT     tcp  --  209.193.93.158       0.0.0.0/0           tcp dpt:443 
ACCEPT     tcp  --  209.193.93.158       0.0.0.0/0           tcp dpt:10000 
ACCEPT     tcp  --  209.193.64.254       0.0.0.0/0           tcp dpt:80 
ACCEPT     tcp  --  209.193.64.254       0.0.0.0/0           tcp dpt:443 
ACCEPT     tcp  --  209.193.64.254       0.0.0.0/0           tcp dpt:10000 
ACCEPT     tcp  --  70.162.22.135        0.0.0.0/0           tcp dpt:80 
ACCEPT     tcp  --  70.162.22.135        0.0.0.0/0           tcp dpt:443 
ACCEPT     tcp  --  70.162.22.135        0.0.0.0/0           tcp dpt:10000 
ACCEPT     tcp  --  209.193.64.2         0.0.0.0/0           tcp dpt:80 
ACCEPT     tcp  --  209.193.64.2         0.0.0.0/0           tcp dpt:443 
ACCEPT     tcp  --  209.193.64.2         0.0.0.0/0           tcp dpt:10000
</file>

=== Redhat Firewall ===

http://wiki.centos.org/HowTos/Network/IPTables

Run iptables commands, then issue:

<file>
service iptables save
</file>
or
<file>
iptables-save > /etc/sysconfig/iptables
</file>

Restore saved rules like this:

<file>
iptables-restore
</file>

===== Night Mode =====

<note warning>This may have changed.  Check!</note>

		Voice Recordings
			Upload two .gsm files to /var/lib/asterisk/sounds/ogm/
				offices-open.gsm
				offices-closed.gsm

===== Dial by Name Directory =====

http://www.asterisk.org/docs/asterisk/trunk/applications/directory

If you want to announce the extension number to the callers, and dial by first name, clone the script tl-directory into tl-directory-first-ext.

To clone a script on Web Interface menu go to PBX Settings > Script Library > choose script to clone and use button at the bottom of the page.

Add '|ef' to the options:

<file>
exten => s,2,Directory(default${TL_DASH}${tenant},local-extensions${TL_DASH}${tenant}|ef)
</file>

Once the script is created, add a feature code that uses the new script.  Then you can use it in IVRs.

===== Root Mail =====

To forward administrative mail to a real user, edit ''/etc/aliases'':

<file>
vi /etc/aliases
</file>

Edit the bottom of the file:

<file>
# Person who should get root's mail
root:           you@your.domain
</file>

Then run the ''newaliases'' command to activate the changes:

<file>
newaliases
</file>

===== Voicemail to E-Mail =====

==== Postfix ====

See also **[[networking:linux:postfix_smarthost|Postfix Authenticated Smarthost]]**

http://www.cyberciti.biz/faq/postfix-smtp-authentication-for-mail-servers/

http://www.dnsexit.com/support/mailrelay/postfix.html

ISO 2.0 install includes the Postfix MTA, so we tweak it to **forward mail to a smarthost** rather than configure everything necessary for a full-blown mail server.

Set the From Address for messages from the PBX:

**System Settings -> Network Settings -> Postfix SMTP rewriting -> iPBX@sonoracomm.com**

==== voicemail.conf ====

Edit ''voicemail.conf''.  Set format for e-mail attachments to 'wav'.

<file>
volgain=6.0					# TDM400P card
serveremail=SonoraPBX
minsecs=5					# was 'minmessage' - adjust to get rid of hangups
fromstring=The Sonora iPBX
emailbody= (change 'Thirdlane' to 'SonoraPBX')
tz=arizona

[zonemessages]
arizona=America/Phoenix|'vm-received' Q 'digits/at' IMp
</file>

===== Digium TDM Cards =====

:!: TDM (FXO) cards don't like to share a hardware interrupt.

Check for shared IRQ:

<file>
lspci -vb
</file>

===== Echo Cancellation =====

Run ''fxotune'' to tune the TDM card to the actual phone lines connected.

  - Connect the phone lines to the FXO ports
  - Shut down Asterisk
  - Run ''fxotune'' to modify the config file
  - Verify the changed config file
  - Restart Asterisk

Shut down Asterisk, then:
 
<file>
fxotune -i 4

cat /etc/fxotune.conf
</file>

Run this to set the running configuration to the settings in ''/etc/fxotune.conf'' (created by fxotune -i 4):

<file>
fxotune -s
</file>

Normally, fxotune -s is run by Zapata startup.

Restart Asterisk.

===== Provisioning =====

==== Polycom ====

http://www.thirdlane.com/forum/ftp

http://www.fonality.com/trixbox/forums/vendor-moderated-forums/polycom/configuring-polycom-phones-tb-2-2-endpoint-mgr

  * Download the appropriate SIP and BOOTROM from Polycom Portal 
  * Unzip both files into /home/PlcmSpIp
    * same location used for HTTP, FTP and TFTP
  * Test with 'ftp pbx'
    * Username 'PlcmSpIp'
    * Password 'PlcmSpIp'
    * > get sip.cfg
    * Listing of files is disabled for security
  * Test with 'tftp pbx' command from another PC
    * > get sip.cfg
  * Test with browser
    * http://192.168.1.30/provisioning/ReadMe.txt

<file>
cd /home/PlcmSpIp/

# For older Polycoms such as 301, 501 and 4000
wget http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_7_legacy_release_sig_combined.zip
wget http://downloads.polycom.com/voice/voip/sp_ss_bootrom/spip_ssip_vvx_BootROM_4_1_4_release_sig.zip
unzip spip_ssip_3_1_7_legacy_release_sig_combined.zip
unzip spip_ssip_vvx_BootROM_4_1_4_release_sig.zip

# For newer phones - don't use the latest as it doesn't work with TL provisioning yet
wget http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_vvx_3_2_5_release_sig_split.zip
wget http://downloads.polycom.com/voice/voip/sp_ss_bootrom/spip_ssip_BootROM_4_3_0_release_sig.zip


chown -R PlcmSpIp.PlcmSpIp *
</file>

==== Snom ====

http://wiki.snom.com/Features/Auto_Provisioning/Redirection

http://www.thirdlane.com/forum/snom-auto-provisioning

===== Polycom Provisioning =====

Polycom Inter-Digit Timeout

==== Firmware ====

Use 'Split' firmware as long as BootRom is v 4.0 or above.

<file>
cd /home/PlcmSpIp/

wget http://downloads.polycom.com/voice/voip/sp_ss_bootrom/spip_ssip_BootROM_4_3_0_release_sig.zip
wget http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_vvx_3_2_5_release_sig_split.zip

unzip spip_ssip_BootROM_4_3_0_release_sig.zip
unzip spip_ssip_vvx_3_2_5_release_sig_split.zip
</file>

=== Files ===

Thirdlane generates these config files based on templates in /etc/asterisk/provisioning

${mac}				(lists config files to be read - static)
${mac}-registration.cfg	(per-phone (line) settings)
local-settings.cfg		(per-installation site-wide settings)
sip.cfg				(static)

Each phone generates this override config file for user-changed settings in /tftpboot:

OVERRIDES/${mac}-phone.cfg

=== Templates ===

Copy /etc/asterisk/provisioning/models.txt to /etc/asterisk/user_provisioning/ and delete all models not being modified.  'user_provisioning' folder will not be overwritten upon TL upgrades and will override 'provisioning' templates.

<file>
cp -a /etc/asterisk/provisioning/models.txt /etc/asterisk/user_provisioning/

vim /etc/asterisk/user_provisioning/models.txt

cp -a /etc/asterisk/provisioning/models.txt /etc/asterisk/user_provisioning/
cp -a /etc/asterisk/provisioning/polycom_* /etc/asterisk/user_provisioning/
</file>

To program the second and third line keys for other functions, specify the number of line keys as 1.

=== Programming ===

http://www.thirdlane.com/forum/custom-polycom-soft-key-programming

Edit /etc/asterisk/user_provisioning/polycom_mac.cfg:

Modify for current version of firmware (change 313 to 317) to fix error 0x20.

Edit /etc/asterisk/user_provisioning/polycom_local.cfg:

Verify that the phone will dial *97 to check voicemail:

<file>
    msg.mwi.1.callBack="*97"
</file>

=== Enable Presence ===

<file>
   feature.1.enabled="1"
   softkey.feature.mystatus="1"
</file>

Use these settings to override DHCP for NTP:

<file>
   <TCP_IP>
      <SNTP 
          tcpIpApp.sntp.resyncPeriod="14400" 
          tcpIpApp.sntp.address="192.168.1.30" 
          tcpIpApp.sntp.address.overrideDHCP="1" 
          tcpIpApp.sntp.gmtOffset="-25200" 
          tcpIpApp.sntp.gmtOffset.overrideDHCP="1" 
          tcpIpApp.sntp.daylightSavings.enable="0"/>
   </TCP_IP>
</file>

Adjust the phone's digitmap as needed.  These settings require hitting 'send' for all calls:

<file>
   dialplan.digitmap=""
   dialplan.impossibleMatchHandling="2"
</file>

=== Testing ===

Format Filesystem on phone to test configuration file changes.

Check Voicemail

<file>
	*97 - feature code - check your own mailbox
	_*85NNNN - Check mailbox NNNN                         ????
</file>

=== Directory ===

Assign fewer lines to phones in order to leave buttons free for programming.

<file>
cp -a /tftpboot/000000000000-directory~.xml /tftpboot/000000000000-directory.xml
</file>

<file>
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<directory>
	<item_list>
                <item>
                        <fn>Park</fn>
                        <ct>700</ct>
                        <sd>1</sd>
                        <rt>1</rt>
                </item>
                <item>
                        <fn>Page</fn>
                        <ctfn>*98</ct>
                        <sd>2</sd>
                        <rt>1</rt>
                </item>
	</item_list>
</directory>
</file>

=== Park and Paging ===

https://www.8774e4voip.com/kb.asp?ID=42

=== BLF ===

http://www.thirdlane.com/forum/shared-line-appearances

Assign fewer lines (1) to leave buttons free for programming in Tools -> Auto-Provisioning.

Polycom - Not yet working for line buttons, only expansion?  May work soon...

Unused line keys are automatically populated with speed-dial contacts based on their speed dial index number. It is the same way you populate the expansion module.

These settings are needed and are default in /etc/asterisk/user_provisioning/polycom_local.cfg:
<file>

feature.1.name="presence"
feature.1.enabled="1"
</file>

Also:

  * create a Contact (First Name)
  * specify the extension (Contact)
  * select yes to 'watch buddy'
  * give it a speed dial index of 1
  * it will be the first listed on the expansion module
  * for these 'watched buddies' go into their extension configuration in pbx manager and limit their calls to 10 calls.
  * Reload asterisk.

==== Thirdlane Dialer ====

Works as a standalone application or as an Outlook toolbar.

http://www.thirdlane.com/downloads/DialerSetup20.exe

http://www.thirdlane.com/products/thirdlane-dialer/configuration

==== vTiger CRM ====

<file>
yum install vtigercrm-ast-en
</file>

==== OpenFire Instant Messaging Server ====

<file>
yum install openfire
</file>
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