VoIP SIP Trunks
A SIP Trunk is a service where by VoIP calls are routed to and from SIP servers. A customer may have their own SIP Server/PBX, and need a 'SIP Trunk' service in order to make and receive calls though AAISP.
Our VoIP system can simple 'trunk' calls to/from your own PBX - it is perfectly normal for you to have your own office PBX, such as a FireBrick or an Asterisk server for example and for your equipment to provide you will all the 'PBX' type functionally that you require. However, you can also have SIP phones register directly against our servers and use the PBX functionality that we provide. Or, indeed, have a mix of the 2 - e.g. we can route calls to you by SIP (trunk), but we can also record calls.
Outbound SIP Trunks
SIP trunks are available from AAISP. In practice all VoIP accounts from AAISP can be used as a SIP Trunk, in that the SIP account can be used to make multiple outbound calls.
There is no limit on the number of outbound calls you can make with an AAISP VoIP account, simply use the username/password and the server in your PBX config.
If you'd like SIP trunks from AAISP, then please do contact the Sales department.
We allow you to present the CLI of any number where the configured base number is one you are using for authentication. The base number is configured under "General" SIP settings for a number.
We are also able to set up an additional number on request, see VoIP Presentation Number.
Inbound SIP Trunks
Inbound calls can be routed to a registered SIP phone, redirected to another number or routed to your own SIP server. To help where a customer may have a block of numbers the AAISP Control Pages have a concept of grouping numbers together to use a common number as their settings.
When we route calls to your own SIP server you do not need to register against our servers. You may if you wish, but we will then send INVITEs to both the registered device and to the SIP server set in the control pages.
In some number ranges (e.g. 033) we offer DDI (direct dial in) blocks. These are blocks of 10, 100, 1000 or even 10000 numbers routed to one destination. You could have several blocks.
To manage the blocks each block has a base number in our control pages e.g. 01234567890 for a block of 10, or 01234567800 for a block of 100. This allows you to control where the numbers are sent (e.g. SIP to your server, registered SIP phone). Incoming calls to any of the numbers is delivered as per this setting.
To make outgoing calls you can use a SIP authenticated login using the base number as the login, and the password you set. This is just the same as using single numbers. One difference is that you can set the CLI (calling number) to any of the numbers in the block. If you set the CLI to one of the numbers then that is what is used. If you do not set the CLI or set an invalid one then the base number is used. If you have a presentation number then this is used instead when you do not set a CLI. This means you can choose to use any of your numbers, or a presentation number if you have one, on a per call basis from your own phone system.
The individual numbers within the block have their own management and billing as normal. This can be useful where most numbers are handled by your phone systems but you need some specific exceptions handled differently.
When a number has a Base set, then many of the settings for that number are hidden - as it will inherit its settings from the Base number.
Then on the Base number, you can set Incoming calls to be set to your server
For example, say you have a block of 10 numbers:
- Set all the 'SIP to Server' settings on the number ending 0
- On the 9 other numbers set the Base/group setting to be the number ending 0
This then makes all the number inherit the settings from the 0 number, i.e., calls for any of the numbers will be sent to the SIP server settings as specified in the Base Number.
It is also possible to register your PBX against the 'Base' Number, then calls for numbers in the 'Group' will be sent to that registration.
Please note that the base number CLI logic can be broken by NAT. We initially look for a registration for the number we have on a call (e.g. DDI), and if found, we use it. If not, and if base number is set, we check if the base number has a registration. If so, we use the DDI followed by the contact from the registration. When NAT is used we can't trust the contact for the registration.
If you are using the asterisk PBX, or systems based on asterisk, please see VoIP Phones - Asterisk for notes on configuration.