Frequently asked questions

A SIP trunk is a voice connection delivered over IP. It links your phone system (an IP PBX or a cloud phone system) to the public telephone network, so calls to and from ordinary phone numbers travel over your internet or private data connection instead of copper or ISDN lines.

Read the full explanation in What is SIP trunking?

Usually, yes. Most modern IP PBXs support standard SIP trunks. Before anything is ordered we review your platform, software version, network and call-routing needs, so any incompatibility (for example an old PBX without SIP support, or a missing codec) is found up front. Older PBXs can often be connected through a SIP gateway.

A channel is capacity for one simultaneous call, in either direction. Ten channels let ten calls happen at the same moment, whether inbound, outbound or a mix. Channels are separate from phone numbers: a business with 50 numbers may need only 8 channels if no more than 8 people are ever on external calls at once. See how to estimate channel requirements.

Not because of the SIP trunk. The trunk connects to your PBX, not to individual handsets, so the phones registered to your PBX keep working. You only need new devices if you are replacing the PBX itself, for example when moving to a cloud PBX that does not support your current handsets.

Yes. A SIP trunk normally carries both directions. Inbound calls arrive on the numbers (DIDs) assigned to the trunk and are delivered to your PBX; outbound calls are sent from your PBX to the trunk for delivery to the destination. Whether specific outbound destinations are enabled, and which caller IDs can be presented, is agreed during setup.

Send us the countries, cities or number types you need (local, national or toll-free) and the quantity. We confirm availability, any documentation the regulator requires (often proof of a local address or business registration), lead time and cost before anything is ordered. Start with a quote request or read about how coverage is confirmed.

Often, by porting them. Whether a number can be ported depends on the country, the number type and your current provider. Porting needs an authorisation from the account holder and matching account details, and timelines are set by the losing and gaining carriers. We check portability before you cancel anything with your existing provider.

Coverage is confirmed per request rather than published as a blanket list, because number availability and calling routes change with regulation and supply. Tell us your destinations and we will confirm what is available. See Service coverage.

Quotes are built from your actual requirements: the number of concurrent channels, the destinations you call, how many numbers you rent and where, the mix of inbound and outbound traffic, expected monthly volume, redundancy needs and any custom integration or compliance work. We explain each line of the quote.

It depends on what is involved. Connecting an already-compatible PBX for outbound calling can be quick once terms are agreed. New numbers, regulated countries and number porting add time that is set by number suppliers, regulators and carriers. We give you a realistic timeline for your specific project before you commit, and we test inbound and outbound calling before go-live.

• Your phone system (platform and version) and whether it is on-premises or hosted
• Peak concurrent calls, or the number of users and how busy they are
• Countries and number types you need, and whether you are porting numbers
• Main calling destinations and an estimate of monthly minutes
• Any redundancy, compliance or integration requirements

Rough numbers are fine. The quote form walks you through them.

The biggest risk with any SIP connection is toll fraud: someone gaining access to your PBX or credentials and placing expensive calls. Good practice includes IP allow-listing, strong unique SIP credentials, keeping the PBX patched and off the open internet, restricting international and premium destinations you do not need, setting channel and spend limits, and encrypting signalling and media (TLS and SRTP) where supported. See SIP security best practices.

With a SIP trunk, calls depend on the data connection between your PBX and the provider. Inbound calls can be configured to fail over to alternative destinations, such as another site, a mobile number or a second trunk, when your PBX is unreachable. Outbound resilience needs a second connection or a second site. We help you plan these routes and test them.

SIP signalling can be encrypted with TLS and audio with SRTP when both your PBX and the route support it. Availability is confirmed per deployment, and we will tell you plainly if a particular configuration cannot be encrypted end to end.