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Peoplefone SIP-TRUNK

PBX / Private branch exchange

Our peoplefone SIP-TRUNK solution is compatible with any private branch exchange (PBX). We support for example 3CX, Asterisk, Aastra Opencom, Cisco, etc.

  • Instructions on how to connect various private branch exchange are available here : PBX
  • You will find an overview of certified PBX for peoplefone SIP-TRUNK on the dedicated page.



If you are using a PBX, the SIP-TRUNK option must be enabled on the SIP line so that the correct information can be transmitted in the SIP Header. This will allow your PBX to correctly process incoming numbers and send calls to the correct destinations.

This activation is free and must be done by your installation partner or by peoplefone.

For international calls, peoplefone supports the international dialling standard according to E.164. Caller ID (Identification of the caller) peoplefone allows two different formats for a phone number. With +41 and the regional prefix (without zero) or the normal phone number (without the country prefix).
Example:+41445522000 or 0445522000.



If you want the number display on outgoing calls to be defined by the PBX, the CLIP OPEN option must be activated on your SIP-TRUNK line. This activation is free and must be done by your installation partner or by peoplefone as indicated here.

Emergency calls

To ensure that emergency calls can be correctly routed to the nearest call centre, some information must be indicated on your customer portal. Please follow the instructions on the Emergency Call setup page.

Encrypted calls

Si vous souhaitez que les appels soient cryptés pour des questions de confidentialité ou pour contourner un SIP-ALG If you want calls to be encrypted for privacy reasons or to bypass an active SIP-ALG on your router or firewall, this is completely possible.

Simply enable TLS and SRTP on your device and contact us by email at [email protected] with the SIP username of the line where it should be enabled.

Individual installation

Simple installation means that you can connect (configure) a device, softphone or APP directly to a SIP line. Using the manual configuration data, you can perform the installation manually via the web interface of the device itself.
Instructions on how to do this can be found below:


You can use different rules to forward your calls to the voicemail by going to Configuration → Call Forwarding Overview and choosing the SIP line to forward.

Customized message

You can use a VoIP device with built-in voicemail. For this, go to the device settings to record the voice message and activate it. Nothing needs to be done in the peoplefone account. With a VoIP device without integrated voicemail you will have to activate the function directly in your peoplefone account as described above. Once the rule is added, dial *1 on your VoIP device, you will need to follow the instructions below to record your message.

  • 0 : voicemail option
  • 4 : temporary message (it will replace the standard message)
  • 1 * speak * # : to record the message
  • 1 : to validate your message