What is a SIP trunk?

A SIP Trunk allow you to route concurrent calls over the Internet to an IP capable PBX system. iTelco uses the SIP protocol to deliver calls to your PBX platform. If you do not have a PBX or VoIP Gateway - or your existing PBX/gateway is not SIP capable then the iTelco SIP trunking service is not for you and you should never enable this feature on your account. SIP trunking can be seen as a low cost replacement for traditional PRI or analog circuits - using the Internet to make/receive calls instead. This means cost savings for you the customer since you do not need to order expensive circuits from your telecommunications provider and can use your existing broadband connection to terminate/place calls instead.

What do I need to setup SIP trunking?

SIP trunking is not for everyone and is primarily used by businesses. To setup SIP trunking you will need the following:

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• A static Internet (IP) Address. Dynamic DNS hostnames are not allowed. You must specify an IP address in iTelco web portal to act as your endpoint for inbound calls.

• No firewall rules or NAT blocking traffic between iTelco servers and your IP-PBX. You may setup firewall rules to limit traffic to iTelco servers or setup port forwarding if required for security reasons. See below for more details.

• You need a SIP (v2) compatible PBX platform or gateway. iTelco does not support H.323 or other VoIP protocols.

If you are unsure whether your IP-PBX is compatible with iTelco then contact support@itelco.nz for more information.

I have to use SIP trunking if I have an IP-PBX?

This depends. Many IP-PBX platforms can register (login) with iTelco as a normal phone would. In fact this is what many Asterisk users do using the 'register' directive in their SIP configuration. If you do not have a static IP address or your IP-PBX is behind NAT then you cannot use SIP Trunking and you will need to register your devices with iTelco and use the 'Inbound trunking' feature on each line instead to achieve a similar goal.

What are the benefits of using SIP trunking?

Many IP-PBX's cannot be configured to register/login to iTelco as a client, so for these types of devices SIP trunking is the only way you can interconnect with iTelco . SIP trunking has many other benefits over a dynamic registration including:

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• Your device does not need to be logged in or registered, so calls will always be routed to your IP-PBX. This means that if your internet connection was reset or went down temporarily you would start receiving calls immediately once your internet connection comes back up (instead of having to wait for a re-registration or login)

• You can route all your numbers easily to your IP-PBX with no further configuration required. If you add more numbers to your account then they will automatically be routed down your trunk.

• You save bandwidth since there is no need for SIP registrations or 'polling' traffic usually sent to devices that are behind NAT.

• SIP trunks have the concept of a 'pilot' number. You can present any caller ID number down your trunk and if it is not recognised as being on your account then the caller ID will be set to the trunk number. This makes configuration simpler and avoids outbound calls ever being rejected.

How do I setup SIP trunking on my iTelco account?

Firstly ensure that your IP-PBX is configured correctly and ready to receive calls from iTelco. Then follow these steps:

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• Decide which iTelco number you wish to act as you 'pilot' number (primary number) on your trunk and login to iTelco web portal with this number

• In the 'iTelco Settings' page under 'Preferences', click on 'SIP Trunking'

• Tick the 'Enable SIP trunking' checkbox

• Enter the static IP address of your IP-PBX in the 'Primary Host IP Address' text box

• If you have a second IP-PBX to act as a 'failover' host then enter this in the next text box (This is an optional field)

• If you do not wish to lose any existing settings on your iTelco lines then tick the 'Do not reset my line settings' box. Usually, a number of features are automatically disabled when you enable SIP trunking on your account such as Voicemail, call transfer features etc. since these features are usually handled by your IP-PBX platform.

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