Once you've got Asterisk, you'll need at least two phones in order to test your phone system. In this tutorial we're going to use three - but don't worry - instead of buying hardware phones, we're going to use free SIP software phones instead, running on another computer on the local network.

We use SJPhone from SJLabs (installation instructions), and X-Lite from CounterPath (installation instructions), and Express Talk from NCH (installation instructions), all running on the same Windows PC. You can use the FREE versions of all these phones to complete the tutorial. We also use MicroSIP and Vop Nano if we need even more phones.

For this tutorial, name your ExpressTalk phone '1001', your X-Lite '1002', and your SJPhone '1003'. Edit your /etc/asterisk/sip.conf file and at the end add the following lines:

                                [1001]
                                ;ExpressTalk
                                context=main
                                type=friend
                                Callerid="1001" 
                                username=1001
                                secret=1001
                                host=dynamic
                                dtmfmode=RFC2833
                                disallow=all
                                allow=alaw

                                [1002]
                                ;XLite
                                context=main
                                type=friend
                                username=1002
                                Callerid="1002" 
                                secret=1002
                                host=dynamic
                                dtmfmode=RFC2833
                                disallow=all
                                allow=alaw

                                [1003]
                                ;SJPhone
                                context=main
                                type=friend
                                Callerid="1003" 
                                username=1003
                                secret=1003
                                host=dynamic
                                dtmfmode=RFC2833
                                disallow=all
                                allow=alaw
                            

If you are using your own sip.conf entries, you must make sure the Callerid and context are set as above to be able to follow the examples in this tutorial.

Hint: If you want to hear Music On Hold with X-Lite, make sure Menu->Advanced System Settings->Audio Settings->Silence Settings->Transmit Silence is set to Yes.

Note: We couldn't get DTMF (keypress) tones from the current version of SJPhone to work at all, so we'll use this as the Caller's phone for the rest of the tutorial. We'll use Express Talk (1001) as the first Agent phone, and X-Lite (1002) as the second.

Add Phones to OrderlyStats

In order for OrderlyStats to start tracking calls to and from your Phones, you need to add the Phones to OrderlyStats. To do this, go to the Agents page (in the menu on the left hand side of the Control Panel) and hit Add Interface at the bottom of the page, and enter in SIP/100X (where X is 1 and then 2 - don't add the third one as this is being used to represent the Callers) for each of the two agent Phones like this:

The username and password fields are for the Agent Toolbar - more about this later.

Note: The LINE Agent is automatically created by the system, and is used to indicate that a call has been answered, but not by an Agent.

Create Phone Extensions

Next, we need to create extensions for your phones, so that Asterisk can dial them.

WARNING If you are running asterisk 1.6.x or above you should first bypass the default AEL dialplan, otherwise your extensions.conf file will be ignored. You can do this by running the following commands as root:

                                > mv /etc/asterisk/extensions.ael /etc/asterisk/extensions.ael.save
                                > asterisk -rx "core restart now"
                            

Note: You must restart asterisk for this change to take effect.

Now add the following to the end of your /etc/asterisk/extensions.conf file:

                                [main]
                                ;Phone Extensions
                                exten => _1XXX,1,NoOp(Dialling phone ${EXTEN})
                                exten => _1XXX,n,Dial(SIP/${EXTEN},,rotwh)
                                exten => _1XXX,n,Hangup
                            

Next do an Asterisk reload by typing module reload (or just reload for Asterisk 1.4 or below) at the Asterisk command line (which you can reach by typing asterisk -vvvr).

Here we have created a new context for your phone calls, called main. Calls made from your phones will arrive in the context specified in each phone's sip.conf entry (which we have already set to main).

We've also defined a pattern-matching extension, which will match any four digit number 1000 to 1999. The extension dials the corresponding SIP phone.

Let's make some calls! To recap, we have set up agent John Doe using phone 1001 (ExpressTalk), and agent Jane Smith using phone 1002 (X-Lite). We're using SJ-Phone to represent your callers.

So, let's use SJPhone to call John Doe, by dialling 1001. You'll see the call come up as an INBOUND call to John Doe in the OrderlyStats Control Panel:

Now we're going to test the second Agent, Jane Smith (1002), using X-Lite.

To test Jane's phone, dial 1002 from John's phone (ExpressTalk). As this is an internal call, each leg of the call will be shown separately in the Control Panel, as an INBOUND call to Jane, and an OUTBOUND call from John:

The call is only shown once in the OrderlyStats reports (historical logs), as an INTERNAL call:

Calls are shown with the most recent at the top of the page (but you can change the sort order using the arrows at the top of each column).

Now you've defined your Agents and their phones, you're ready to set up a queue.

Queue definitions are found in /etc/asterisk/queues.conf. As you'll see there are a lot of configuration options; here are two example queues to get you started.

                                [Sales]
                                music=default
                                strategy=ringall
                                eventwhencalled=yes
                                timeout=15
                                retry=1
                                wrapuptime=0
                                maxlen = 0
                                announce-frequency = 0
                                announce-holdtime = no

                                [CustServ]
                                music=default
                                eventwhencalled=yes
                                strategy=ringall
                                timeout=15
                                retry=1
                                wrapuptime=0
                                maxlen = 0
                                announce-frequency = 0
                                announce-holdtime = no
                            

These queues will ring all assigned Agents when a new Caller arrives, and will not make any announcements regarding queue size, estimated wait etc.

If you've defined a custom music class, you can set your queue to use it with the music parameter.

Do an Asterisk reload, and the new queues will appear in the Control Panel within 10 seconds:

It is possible to assign static Agents to your queues in queues.conf with a member line like this:

However, this is not a good idea as static Agents cannot then be logged in and out of queues dynamically - they are permanently assigned instead. This means that static Agents have the following drawbacks:

• Static Agents cannot log in and out when they arrive for work and leave for home,

• Static Agents' session times cannot be tracked,

• Callers cannot be automatically routed away from the queue if no-one's there to answer them,

• Static Agents cannot be dynamically reassigned to different queues, and

• Static Agents' phones are rung all the time, regardless of whether the Agent is present to answer, so you will have a room full of ringing phones all the time.

These drawbacks are removed by using dynamic Agents instead as described in the next section.

Hint: You can use Group functionality to assign groups of Agents to queues, but many of our users have had problems with this system, so we don't recommend it. It's much better to add Agents individually to each queue.

Hint: If you have already set up your Asterisk queues, you should remove any 'member =>' lines from your queues.conf, otherwise your agents won't be able to dynamically log in and out of your queues. You can manage agent assignment in the OrderlyStats Agents page.

To recap, so far we have defined our Agents and the phones they use.

We have also defined two queues, so the last thing we need to do is create the mechanism that allows Agents to log in and out of the queues, so they can start receiving queued calls.

The Hard Way

Skip to The Easy Way if you're using OrderlyStats.

If you're not using OrderlyStats, you can provide your Agents with numbers to dial to log in and out of individual queues, but you must then give your queues numeric names. Here is an example:

                                [macro-queue-login]
                                exten => s,1,Set(agent=${ARG1})
                                exten => s,n,Set(queue=${ARG2})
                                exten => s,n,NoOp(Queue login agent ${agent} to queue ${phone});
                                exten => s,n,AddQueueMember(${queue},SIP/${agent});
                                exten => s,n,Playback(agent-loginok)

                                [macro-queue-logout]
                                exten => s,1,Set(agent=${ARG1})
                                exten => s,n,Set(queue=${ARG2})
                                exten => s,n,NoOp(Queue logout agent ${agent} from queue ${phone});
                                exten => s,n,RemoveQueueMember(${queue},SIP/${agent});
                                exten => s,n,Playback(agent-loggedoff)


                                [main]
                                ;...

                                exten => _3XXX1,1,Answer
                                exten => _3XXX1,n,Read(Agent,Agent-user)
                                exten => _3XXX1,n,Macro(queue-login,${Agent},${EXTEN:0:4})
                                exten => _3XXX1,n,Hangup

                                exten => _3XXX11,1,Answer
                                exten => _3XXX11,n,Read(Agent,Agent-user)
                                exten => _3XXX11,n,Macro(queue-logout,${Agent},${EXTEN:0:4})
                                exten => _3XXX11,n,Hangup
                            

This has the following limitations:

• All your queue names must be four digits long and start with the number 3 (which is not user-friendly), and

• The Call Center Manager cannot manage which Agent is assigned to each queue, and,

• Agents who are members of multiple queues will have to log in and log out of each one separately, and

• Agents may not be able to perform login or logout if their phone is already ringing or in use, and

• There is no support for Agent prioritisation (using the Penalty system), and

• There is no support for wrap-up (using the Pause system).

The Easy Way

The easy way to do queue login and logout is with OrderlyStats. OrderlyStats comes with an Agent Toolbar that your Agents use to log in and out of their queues. The first step is to assign the Agents to the queues, using the Agents page - click on the 'None' link at the end of each Agent's row to open the assigner dialog, like this:

If you're not using OrderlyStats, you can provide your Agents with numbers to dial to log in and out of individual queues, but you must then give your queues numeric names. Here is an example:

Although the Agents are now assigned, they won't start taking calls until you log them in with the Login button (Login Button), which you can do from this page as Call Center Manager.

Agent Toolbar

For your Agents to log themselves in and out of your queues, they need the Agent Toolbar. You should have already set them up with a username and password for this in the 'Add Phones to OrderlyStats' section above. Once logged in, the toolbar looks like this:

To Log Out of all her queues, Jane just needs to hit the Log Out button. She can log in to all her assigned queues with one touch by hitting the Log In button.

If your Agents need time to wrap up after each call before taking the next call, they can do this by hitting the Pause button during the call, or aftwerwards. This will prevent the queue(s) to which they are assigned from sending them more calls until they hit the Resume button.

The OrderlyStats Reports will show you Agent session times, and also time spent in wrap up after each call.

Note: The Toolbar Login/Logout function logs an Agent in or out of all the queues to which the Agent is assigned in the Agents page.

Agents do not have direct control over which queues they are assigned to with the Agent Toolbar. This is by design.

As Call Center Manager, you can do this yourself at any time using the Agents page to reassign Agents on the fly, or log them in and out yourself using the Log In/Log Out buttons next to each Agent in the Control Panel. You retain full control of which queue(s) each Agent is assigned to.

Hint: Some versions of Asterisk contain a bug that prevents agents from initialising correctly when Asterisk is restarted. You can check to see if your version of Asterisk is affected by restarting Asterisk, and then running queue show or show queues from the command line. If your agents are shown as Invalid, then you can easily fix this by adding the following lines to the preload section of your /etc/asterisk/modules.conf file:

                                preload => pbx_config.so
                                preload => chan_local.so
                                preload => chan_sip.so
                            

The last step is to provide an extension for incoming Callers to ring into the queue. Add the following lines to /etc/asterisk/extensions.conf at the end of the [main] context.

                                exten => 3001,1,Answer
                                exten => 3001,n,Ringing
                                exten => 3001,n,Wait(2)
                                exten => 3001,n,Set(_doVoicemail=no)
                                exten => 3001,n,Queue(CustServ,twh)
                                exten => 3001,n,Hangup

                                exten => 3002,1,Answer
                                exten => 3002,n,Ringing
                                exten => 3002,n,Wait(2)
                                exten => 3002,n,Set(_doVoicemail=no)
                                exten => 3002,n,Queue(Sales,twh)
                                exten => 3002,n,Hangup
                            

Next do an Asterisk reload.

To test your queue, use SJPhone (1003) to dial extension 3001 (CustServ).

You should hear music on hold through SJPhone, and the other phones will start ringing.

You will see something like this in the Control Panel:

Note that if there are no Agents available, the Caller will hear music on hold indefinitely.

Hint: If you want to change this behaviour, then add leavewhenempty=yes to the queue's entry in queues.conf, and replace the Hangup line in the above example with, for instance, a call to leave Voicemail instead. As of Asterisk 1.2.0, this is the default behaviour.

Steps 1-7 will give you a call centre with a traditional ACD / on hold queue system.

However, the traditional ACD as implemented in Asterisk is not optimised, so you will not be answering as many callers as you should when using an optimised queue.

To answer all your callers, you will need our OrderlyQ add on, which guarantees you a > 99% answer rate, and at least a 300% return on investment.

The OrderlyQ system also means that your callers will not have to wait on hold in order to get through, so you will see massive increases in caller satisfaction, as well as a major improvement to your business bottom line.

To get started with OrderlyQ, just request a demo and we'll tell you all about it.