Let’s talk about PBXes

Baby.

So let’s say you’ve got a phone bill that’s quite large, so you want to convert your vast number of analog phone lines to VOIP, except for the one analog line you’re keeping around in case anything goes horribly pear-shaped.

There’s a product out there called MyPBX Soho which will do that for you, unless you want to also record phone calls for Training Purposes, in which case you have to shell out $3,000 for a MyPBX Enterprise, which you’re not going to do because you’ve developed cheapness as some kind of innate sixth sense.

Why not set up an Asterisk server on a virtual machine, hook that up to your VOIP provider, get a MyPBX Soho box to handle your analog calls, and start raking in the cash, assuming you pay your phone bills in cash and you own a rake.

What do I need ?

• First of all, pick up a MyPBX Soho server[1] (USD$281.03 on ebay, you could probably get them cheaper).
• You’ll also need an FXS/FXO Module (USD$77.68, also on ebay). This is a little circuit board that clips into the MyPBX box and allows you to plug it into both a landline phone (FXS) and the hole in the wall (FXO). You can get more FXS modules or FXO modules if that’s the sort of thing you want to do, or a GSM or UMTS module if you want to slot in a SIM card to create a mobile phone that never moves.
• You’ll also need a router that supports PoE to power the phones, so let’s say a TP-Link TL-SF1008P from Umart ($59.00)
• Then you’ll need some Yealink T26P VOIP handsets from ebay ($99.00 each, Free P&P).
• If you don’t have one, grab an analog phone, preferably one that still operates when the mains power is out (i.e. has a handset connected via a cable).
• Purchase a wdpvoip $15.00/month pre-paid business plan, and throw another $7.95 a month at them for a phone number that looks like a phone number. This will give you 15 concurrent calls, with 10c/call to landlines and 18c/minute to mobiles. Feel free to shop around.
• You’ll also need an asterisk server. I opted for the FreePBX web frontend running on Ubuntu, running in a VMWare Workstation hypervisor.
• Extraordinary amounts of spare time

What you want to do, apparently, is to set up a PBX in which asterisk will handle all calls (so that everything is recorded) predominantly over the VOIP provider to save cash, but has bypasses in place if you want to use the analog line, what with your internet provider being as flaky as an exzhema patient from flakeberg who just dropped out of the flake academy and started a diet consisting of nothing but cadbury flakes. That flaky.

Anyway.

I dub thee bnepbx01

Once you’ve got all the bits and pieces listed above, you can start plugging things in, assigning names and numbers to things, putting stickers onto those things so you don’t forget what their names are, and ensure that you can communicate to everything on the network. These days I generally name things in the form data-centre-TLA+closest-IATA-airport-code+what-the-hell-it-is-TLA+(sequence number), omitting the data centre prefix for anything that I can physically touch with my own hands.

My initial setup, with two digital handsets and one analog handset, will look like this:

The imaginary phone numbers I’ll be using below are

(Because I know nothing about telephones, I’m going to call a standard telephone handset and line ‘analog’, and anything that goes over the internet ‘digital’).

Setting up FreePBX / Asterisk (bneast01)

For the Asterisk server, download a 32-bit Ubuntu 14.04.1 (trusty) LTS image, fire that up with about 16GB HDD and 1GB RAM.

Then run these commands (mostly sourced from the instructions at Installing FreePBX on Ubuntu 12.04 Server (Precise Pangolin) ):

sudo apt-get install openssh-server
cd /usr/local/src
sudo wget https://iksemel.googlecode.com/files/iksemel-1.4.tar.gz
sudo tar xzf iksemel-1.4.tar.gz
cd iksemel-1.4/
sudo ./configure
sudo make
sudo make install
cd /usr/local/src
sudo wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
sudo tar xvzf dahdi-linux-complete-current.tar.gz
cd dahdi-linux-complete-2.10.0+2.10.0/
sudo make all
sudo make install
sudo make config
cd /usr/local/src
sudo tar xvzf libpri-1.4-current.tar.gz
cd libpri-1.4.15/
sudo make
sudo make install
cd /usr/local/src
sudo git clone http://git.freepbx.org/scm/freepbx/framework.git freepbx
export VER_FREEPBX=2.11
cd freepbx
sudo git checkout release/${VER_FREEPBX}
sudo adduser asterisk --disabled-password --gecos "Asterisk User"
sudo chown asterisk. /var/run/asterisk
sudo chown -R asterisk. /etc/asterisk
sudo chown -R asterisk. /var/{lib,log,spool}/asterisk
sudo chown -R asterisk. /usr/lib/asterisk
sudo mkdir /var/www/html
sudo chown -R asterisk /var/ww
sudo chown -R asterisk /var/www
sudo sed -i 's/(^upload_max_filesize = \).*/\120M/' /etc/php5/apache2/php.ini
sudo sed -i 's/\(^upload_max_filesize = \).*/\120M/' /etc/php5/apache2/php.ini
sudo cp /etc/apache2/apache2.conf /etc/apache2/apache2.conf.orig
sudo sed -i 's/\(User\|Group\).*/\1 asterisk/' /etc/apache2/apache2.conf
sudo service apache2 restart
sudo vi /etc/apache2/apache2.conf
sudo cp /etc/apache2/apache2.conf.orig /etc/apache2/apache2.conf
sudo sed -i 's/^\(User\|Group\).*/\1 asterisk/' /etc/apache2/apache2.conf
sudo service apache2 restart
export ASTERISK_DB_PW=abc123
mysqladmin -u root create asterisk -p
mysqladmin -u root create asteriskcdrdb -p
mysql -u root asterisk -p < SQL/newinstall.sql
mysql -u root asteriskcdrdb -p < SQL/cdr_mysql_table.sql
mysql -u root -p -e "GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY '${ASTERISK_DB_PW}';"
mysql -u root -p -e "GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY '${ASTERISK_DB_PW}';"
mysql -u root -p -e "FLUSH PRIVILEGES;"
sudo ./start_asterisk start
sudo ./install_amp --webroot /var/www
sudo amportal a ma installall
sudo amportal a reload
sudo ln -s /var/lib/asterisk/moh /var/lib/asterisk/mohmp3
sudo amportal start
cd /usr/lib/asterisk/modules/
sudo wget http://asterisk.hosting.lv/bin/codec_g729-ast110-gcc4-glibc-pentium4.so
sudo chown asterisk:asterisk codec_g729-ast110-gcc4-glibc-pentium4.so
sudo chmod a+x codec_g729-ast110-gcc4-glibc-pentium4.so
sudo /etc/init.d/asterisk restart
sudo asterisk -rx 'core show translation'
sudo asterisk -rx 'sip show channels'
sudo asterisk -rx 'sip show peers'
sudo asterisk -r -vvvv

That last command will put asterisk into remote verbose mode, so you'll be able to see on the command-line what's currently not working properly. There's also logs in /var/log/asterisk.

Now that you've got FreePBX, Asterisk and the G729 codec installed, try connecting via a web browser to your FreePBX installation, to MyPBX and to your phones, update your firmware, set up your SIP accounts, trunks, extensions and incoming/outgoing routes, and then everything will work first time honest.

Despite FreePBX and MyPBX both being based on Asterisk, you'll find they have slightly different terminology for the same thing, so that will make things more entertaining.

Huh?

Here's a network diagram describing the proposed configuration, using industry-standard white-puffy-clouds to depict anything you don't have direct and complete control over.

Select the big chunky buttons on the side to view the different trunks and routes you'll be configuring for this thing.

The Network Diagram

Wha?

With that in mind, here's the exciting details.

You'll probably want to check the different systems as you configure things to make sure the connections vaguely work.

MyPBX configuration

Pry the top off the MyPBX to expose the slots inside (outlined in red). My FXO/FXS card is in the first slot, so port 1 on the back is the analog handset, and port 2 plugs into the phone socket on the wall.

Put the cover back on, connect the ethernet, phone and power cables and turn it on. By default the MyPBX box is on a weird subnet (192.168.5.0/24) [2], so change your laptop IP address to something on the same subnet (e.g. 192.168.5.1), connect to http://192.168.5.150, log in (as admin / password), change the network settings by clicking the 'System' link up the top, then 'LAN Settings' on the side, save the changes then change your laptop IP address back.

When you log back into the MyPBX console (for me, on http://192.168.0.123 ), turn on the debug log (Reports->System Log->Enable Hardware, Normal, Debug and Web logs) and the SSH server (System->Security Center->Service->SSH->Setting), and log in via SSH (192.168.0.123 port 8022, username/password = root/ys123456) and tail -f /var/log/yslog/debug.log, which will be useful when things inevitably don't work.

Back in the web console, click 'PBX' -> 'FXS/VOIP Extensions'. You'll find some existing VOIP extensions already set up (ext 300-305) which you can ignore. Click 'Add extension' and create the new analog extension 103 with the settings:

Name	103
Caller ID	103
Hook flash detection	1000ms (default)
Pickup group	--- (default)
Max call duration	empty (default)
Screenshots

Then click 'PBX' -> 'Physical Trunk', and you should see a pstn2 trunk already set up.

Add a 'Service Provider' called bneast01Trunk with the settings:

Name	bneast01Trunk
Type	SIP
Hostname/IP	192.168.0.120:5060
Maximum Channels	0
Codecs	G722/G726/alaw/none/none
Transport	UDP
Qualify	checked
DTMF Mode	rfc2833
DOD Settings	mostly blank
Associated Extension	103
Screenshot

This trunk will handle communications over SIP for VOIP traffic from the asterisk server bneast01.

Click 'Outbound Routes' and add the following routes. MyPBX has a 15-character limit for outgoing route names, so I had to condense my initial names for these things into an even more incomprehensible form. You'll need routes for outgoing analog (extToPstn2RO), digital (extToBneast01RO) and external incoming (pstn2ToBneast01) calls:

Route name	extToPstn2­RouteOut	extToBneast01Trunk­RouteOut	pstn2ToBneast01Trunk­RouteOut
15-char route name	extToPstn2RO	extToBneast01RO	pstn2ToBneast01
Password	empty (default)	empty
T.38 Support	No (default)	No	No
Rrmemory Hunt	No (default)	No	No
Office Hours	Empty (default)	Empty	Empty
Dial Pattern 1	9.	1.	.
Strip 1	1		1
Prepend 1	empty	empty	600
Dial Pattern 2		0.
Strip 2
Prepend 2		empty
Member Extensions	103(FXS)	103(FXS)	300(SIP) **
Member Trunks	pstn2(FXO)	bneast01Trunk(SPS)	bneast01Trunk(SPS)

**- this extension isn't used, but MyPBX requires something in here, so put in anything.

Note the two dial patterns on bneast01Out; the first is for calls to other extensions (which in my configuration all start with '1'), the second is for outgoing external calls (which in my configuration require the user to dial '0' to get an outside digital line).

Inbound routes on the MyPBX also configure the handling of those routes:

Route name	pstn2ToBneast01Trunk­RouteIn	bneast01TrunkToPstn2Out­RouteIn	bneast01TrunkToExt103­RouteIn
DID Number	empty (default)	9.	103
Extension	empty (default)	empty	empty
Caller ID Number	empty (default)	empty	empty
Distinctive Ringtone	empty (default)	empty	empty
Enable callback	No (default)	No	No
Member trunks	pstn2(FXO)	bneast01Trunk(SPS)	bneast01Trunk(SPS)
Business Days
Office Hours Destination	Outbound Routes / pstn2ToBneast01	Outbound Routes / extToPstn2RO	Extension / 103
Non-office Hours Destination	Outbound Routes / pstn2ToBneast01	Outbound Routes / extToPstn2RO	Extension / 103
Screenshot

Asterisk / FreePBX configuration

Log in to FreePBX at http://192.168.0.120 using the username / password you created earlier, if you did, which I forget.

Go to Applications->Extensions and add the following extensions:

Display Name	101	102
secret	123456	123456
dial	SIP/101	SIP/102
mailbox	101@device	102@device
Voicemail/password	101	102
Voicemail/email	knoxg+101@randomoun.com	knoxg+102@randomnoun.com
Screenshot

Then go to Applications -> Ring groups and add the following ring group:

Group description	allRingGroup
Ring strategy	ringall
Extension List	101 and 102
Destination if no answer	Voicemail (101)
Screenshot

Go to Connectivity->Trunks and add the following trunks:

Description	bnepbx01Trunk	wdpvoipTrunk
CID Options	Allow Any CID	Allow Any CID
Maximum Channels	empty	empty
Dialed Number Manipulation Rules	empty	empty
Outgoing Settings
Trunk Name	bnepbx01TrunkOut	wdpvoipTrunkOut
PEER Details	host=192.168.0.123 type=friend allow=alaw context=from-internal trunk=yes insecure=invite	username=nnnn*nnnnn type=peer secret=yourWdpvoipSecret qualify=yes pedantic=no insecure=very host=27.50.78.2 fromuser=nnnn*nnnnn fromdomain=27.50.78.2 disallow=all allow=alaw
Incoming Settings
USER Context	bnepbx01TrunkIn	nnnn*nnnnn
USER Details	host=192.168.0.123 type=user context=from-trunk	username=nnnn*nnnnn type=user secret=yourWdpvoipSecret qualify=no insecure=very fromuser=yourWdpvoipSecret context=from-trunk canreinvite=no
Registration
Register String	empty	nnnn*nnnnn:yourWdpvoipSecret@@ 27.50.78.2/nnnn*nnnnn
Screenshots

(The nnnn*nnnnn and yourWdpvoipSecret values should be supplied to you by wdpvoip during signup. The register string should all be in one line; there's also some additional instructions on their site )

Go to Connectivity->Inbound routes and add the following routes:

Description	wdpvoipRouteIn	allRouteIn
DID Number	nnnn*nnnnn	empty
Call Recording	Allow	Allow
Set Destination	Ring groups / allRingGroup	Ring groups / allRingGroup
Screenshot

And Connectivity->Outbound routes

Description	defaultRouteOut	wdpvoipRouteOut	bnepbx01RouteOut
Dial pattern 1	( ) + | [NXXXXXXX / ]	( ) + 0 | [X. / ]	( ) + | [103 / ]
Dial pattern 2	N/A	N/A	( ) + | [9. / ]
Trunk sequence 0	wdpvoipTrunk	wdpvoipTrunk	bnepbx01Trunk
Screenshots

Yealink digital phone configuration

The phones will get an IP address via DHCP by default, so press the OK button on the front to find out it's IP address then login to it by browsing to http://(ip-address) and entering the username/password admin / admin

Give it a static IP address, using the following settings under Network -> Basic (or use DHCP if you feel like configuring your DHCP server instead).

Apply the settings, then login to it again on the new IP.

Download the latest firmware from the Yealink site, go to Settings -> Upgrade, and upload the new firmware to the phone.

Once the phone reboots, login again, go to the Account tab, and create an Account for the phone so it can contact the asterisk server on bneast01.

Extension	101	102
Line Active	Enabled	Enabled
Label	101	102
Display Name	101	102
Register Name	101	102
User Name	101	102
Password	123456	123456
SIP Server 1
Server Host	192.168.0.120	192.168.0.120
Screenshot		Pretty much the same as 101, except everything's called 102 instead

Under Features -> General information, upload a custom logo if you like. You'll need a black and white bitmap for the image, and YealinkDOB.exe (requires Windows 7), written by a YeaLink forum member, which converts PNG and BMP files into the proprietary .DOB format used by the handset. You can ignore the conversion program supplied by the manufacturer, which it turns out is an absolute crock of shite and doesn't produce files that the handset will accept.

Analog phone configuration

Plug it in

Spend a while trying to get that to work

Obviously, none of this will work initially, so make a list of things that you expect to work and then cross them off as you spend ages sifting through the logs, trying different codecs, and slightly tweaking settings until you get something that vaguely works, and then Never Touch It Again.

Final Checklist

To give some idea what this configuration will actually do, this is the checklist that I used to whilst setting this up:

And, if you can check all those, then you can spend an astonishing amount of time writing that up.

Cheerio then.

Some reference material:

• MyPBX Soho User Manual
• Intercommunication between two MyPBX (via VoIP Trunking)
• Trunking FreePBX with Yeastar MyPBX
• FreePBX: Adding extensions
• FreePBX: Extensions Module - SIP Extension
• FreePBX: Call Recordings Module User Guide
• Asterisk: Asterisk Dialplan Patterns