Course Content - Basic
Welcome to the online training series from 3CX. In this module we will be explaining how to use a SIP Trunk in your 3CX installation.
To start off we will be discussing the Prerequisites and what is required in order to successfully connect to the PSTN.
Then we will move on and explain the 2 types of Providers which are available.
Finally, we'll discuss what a SIP Trunk is, how to implement it in the Management Console, how to configure Inbound Rules, Outbound Rules and also how you can manipulate the Caller ID that users will present, when making outgoing calls.
What you will need in order to do what is explained in this module is quite simple.
First you will need an account from one of our supported VoIP providers. You can find a list of all our supported providers at the link shown on your screen right now.
Also, as we are going to be setting up a VoIP Provider, you must make sure that you have configured your firewall accordingly and that the Firewall Checker passes. More information about firewall configuration can be found in the Firewall Configuration Module.
We recommend that you use a supported VoIP Provider as they have been tested with 3CX. All supported providers have passed 3CX’s InterOperability Test, or IOT as we call it, which means the provider’s infrastructure is 100% compatible with our software and they have also committed to jointly support the solution.
We have made a template which auto configures the supported trunk for 3CX and should any issue occur, our Support Team will be able to assist you.
The ability to process faxes through the T.38 protocol is not a compulsory requirement for the provider to be considered supported.
You do, however, have the option of using a non-Supported provider, and assuming they comply with RFC 3261, they should work.
Note that if you choose to go down this route, 3CX will not offer support, we cannot guarantee the smooth operation nor do we provide a template. This means that you may have to do some manual configuration within the SIP Trunk settings to get everything to work correctly.
This requires SIP knowledge and time. If you wish to use a particular unsupported provider we recommend you ask them to apply for the 3CX IOT.
What is a SIP Trunk though?
SIP Trunks nowadays have become a standard in the telecommunications world. They offer an alternative way of getting a PSTN Number, but rather than using the old analogue network, they operate over an existing internet line.
No additional hardware is required.
And, you have the option of getting international numbers terminating on your PBX. This enables you to pay local rates for outbound calls and save your international customers money when calling you from abroad.
In order to add a Supported SIP Trunk to your 3CX installation, first log into your Management console and go to the “SIP Trunks” page on the left, then press the “Add SIP Trunk” button.
From the list that is presented, select your country and then select the Supported Provider you have an account with.
Finally set the main trunk number as it has been given to you by the Provider. This will act as a catch all for any non configured DIDs.
Speaking of DIDs, with SIP Trunks, apart from your Main number, you can also get additional numbers that are associated with the main number you already have.
These are called DIDs, or Direct Inward Dial numbers.
DIDs provide your Callers with additional numbers they can dial to reach your system. This enables 3CX to route incoming calls differently, depending on which number was dialed by the Caller.
You can route calls to any of your Extensions, to queues, to other external numbers or even to the built-in Fax Server.
You can change the routing of inbound calls depending on what time of the day it is, or even based on whether it is a public holiday or a weekend.
So far we have shown you how to create a SIP Trunk and we have talked about what DIDs are.
Now we are going to show you how to associate your DIDs with your SIP Trunk in 3CX.
Open the Management Console and go to SIP Trunks, highlight your SIP Trunk, press “Edit” and go to the “DIDs” tab.
Now click “Add” to add your DID numbers. Don’t write the whole number. Instead, use the * character as a wildcard, followed by the last 6 digits, at least, of each DID.
This will ignore all numbers before the last 6 digits of the DID. We do this so the PBX can recognise the DID in whatever format the provider may be sending it. SOmetimes the DID will be sent from the provider in E164 format. Sometimes it may be in international format, or even in local format. With the wildcard, the PBX will ignore the beginning on the number and focus on the last 6 digits only.
Now that we have created our SIP Trunk, and associated any DID numbers we have with it, it is time to configure how the incoming calls will be routed.
To do this, in the Management Console, go to Inbound Rules and press “Add DID Rule”. At this point you will be prompted to fill in some fields.
First, the Name of the Inbound Rule. This is useful, because whenever a user of the system receives a call, they see this name on the display of their device. This way they will know which Inbound Rule was used, and answer the call accordingly. It is, however, optional to add the name, but does help, especially when an extension is assigned various DIDs.
Then, from the DID/DDI drop-down menu, select which DID number that will be used for this rule. These are taken from the DIDs tabs of the SIP Trunks created on the PBX. Only unassigned DIDs will show here.
Finally, choose where the call is going to be routed to during in office and out of office hours, and press OK.
The next thing we will walk you through is how to make outbound calls using your newly created SIP Trunk.
To do this you must create Outbound Rules. Outbound Rules give the administrator a way to control how a call will be routed. This can be done based on the dialed number, the extension trying to make the call, the extension group the dialing extension is in, or even the length of the dialed number.
In this module we will not expand on all the Outbound Rule Options and capabilities. This is covered in the Advanced Outbound Routing module. For now, we will show you how to create a single Outbound Rule that will allow every user of the system to use the SIP Trunk.
Go to the “Outbound Rules” page on the left hand menu of the Management Console, and press the “Add” button. First, set a name for your Outbound Rule. Unlike the Inbound Rules, the name for the Outbound Rules is mandatory. Then, in the “Calls to numbers starting with prefix” criterion field, enter “0-9,+”. Essentially, with this, we are telling 3CX to allow any calls starting with any dialed number or the + character.
After the criteria of the outbound rules have been matched, we must tell 3CX which Trunk to use in order to make the call. This is done by setting Route 1 to the Provider we want.
Once we press OK, the Outbound Rule will be created and any user on our system will be able to make outbound calls through the trunk we specified.
If the provider in the 1st route is busy, or not available, the PBX will attempt to process the call using the subsequent routes.
Emergency numbers are the telephone numbers for the Police, Ambulance or Fire Department. Usually the emergency numbers are 3 digit numbers, which may overlap with the extension numbering.
During the deployment of the PBX, you will need to identify the emergency numbers which will be used. Avoid configuring extensions with these numbers. Common emergency numbers are: 112 in Europe, 911 in the United States, 999 in the United kingdom and 000 in Australia.
Emergency numbers are configured from within the PBX settings. This will define a default outbound rule, which will be defined as the entire emergency number being the prefix.
Calls to emergency numbers can also be limited to particular extensions or extension groups, and this is configured within the emergency number settings.
Calls to emergency numbers can bypass the license restriction temporarily, to allow the call to go through without hindrance.
Moving on to the Outbound Caller ID. The Caller ID is the number the PBX sends out to the PSTN and the receiver of a call sees on their display.
3CX offers 2 ways of manipulating the Caller ID.
The first way is to configure 1 Caller ID that will be used from all extensions when making an Outbound Call.
To configure this, go to your SIP Trunk and Edit the trunk.
Go to the “Caller ID” tab, and in the “Outbound Caller ID” field enter the Caller ID you want to present.
Alternatively, you can configure the system so that each Extension has a different Caller ID when calling out.
To do this, go to the “Extensions” page, highlight an extension you want to set the Caller ID and press “Edit”. Then, in the “General” tab set the Caller ID you want this extension to present when calling out.
You will also need to define this in the Outbound Parameters of the SIP Trunk, so the PBX knows which of the two values to use.
The Caller ID number you enter will usually need to have a specific format. This format can vary from one Provider to another, so we would recommend calling your Provider, and asking them what their infrastructure requires, or look in their documentation provided to you when registering for their service.
Here you can see some of the most common formats providers expect, but we have added spaces for emphasis. Do not add spaces or dashes when defining Caller IDs.
Finally, the last topic we are going to be covering in today’s module is how to recognize some of the more common issues related to SIP Trunks.
If you ever notice that incoming or outgoing calls work, but drop after about 32 seconds or that there is one way audio, specifically when the External Party can hear a user of the system, but the user of the system cannot hear the External Party, this most likely means that your Firewall has not been correctly configured, so double-check that the Firewall Checker has passed and that SIP ALG has been disabled.
Also check if your Public IP has changed from that which you had entered during the initial setup of the system. An incorrect public IP will cause the provider to attempt communication with the wrong public IP. As the PBX will not be receiving any communications with the provider, it will drop the call.
Thank you, and goodbye!