SIP trunking for business: connect your PBX or platform

Set up authentication, transport and codecs with a guided assistant, then read a clear diagnosis when a call does not go through. Custom quote pricing.

Custom quote

A SIP trunk links a private branch exchange (PBX) or telephony software to the phone network over the internet. It replaces physical lines with a logical connection that carries several calls at once.

The setup assistant walks you through authentication (allowed IP addresses or credentials), transport, the codecs offered and the concurrency limit. You can also define a failover destination.

When a call fails, a readable diagnosis explains what was negotiated and where the exchange stopped. Technical details stay available in an expandable panel.

How it flows

Configure
Authenticate
Negotiate codecs
Call established
Failover if needed

Practical uses

Company PBX

Connect your existing exchange without changing your phones.

Contact center platform

Connect a dialer or customer relationship software to your numbers.

Voice agent

Send calls to a voice agent platform over SIP.

Multiple sites

Use one trunk per site or project and follow each one's usage.

Features

  • Authentication by allowed IP addresses or credentials
  • Choice of transport and codecs
  • Concurrency limit per trunk
  • Failover destination when your system is unavailable
  • Display of the capabilities actually negotiated
  • Readable diagnosis with expandable technical details

Limits to know

  • Accepted transports, codecs and authentication modes depend on the selected provider and your contract.
  • Negotiated capabilities can differ from the ones requested. The assistant shows what was actually retained.
  • The concurrency limit is set by contract. Calls beyond it are refused.
  • SIP is offered only in the countries and for the destinations validated with the provider.
  • Call quality also depends on your local network and equipment.
Fictional configuration, for illustration. Addresses, names and values do not match any real service.
# SIP trunk, fictional example
name               : trunk-project-a
server             : sip.example.test
transport          : TLS
authentication     : allowed IP addresses
  address 1        : 203.0.113.10
  address 2        : 203.0.113.11
codecs (in order)  : opus, PCMA, PCMU
concurrent calls   : set by contract
failover           : +33199001235
attached numbers   : +33199001234

Technical details

AuthenticationAllowed IP addresses or credentials
SettingsTransport, codecs, concurrency, failover destination
DiagnosisReadable summary and expandable technical details
AttachmentNumbers and trunk tied to a project

Frequently asked questions

Which codecs and transports are accepted?

They depend on the provider selected for your country and on your contract. The assistant lists only the options that are actually available to you.

What happens if my PBX is unavailable?

If you defined a failover destination, the inbound call goes there. Without one, the call fails and the event appears in the log.

Can I move my current numbers onto the trunk?

Yes, through porting, depending on the country and the current carrier. The preparation guide lists the documents usually requested.

Related use cases

Documentation

Related services

Let's connect your PBX

Tell us about your equipment, the number of simultaneous calls you expect and the countries involved. We will prepare a suitable quote.