
Trunks are used to connect your PBX to external VoIP systems or devices, enabling outbound and inbound call traffic between networks.
The PBX supports two types of VoIP trunks:
User Trunk: A registration-based SIP trunk that requires authentication (SIP REGISTER) using a username and password. Typically used when connecting to a VoIP provider or another PBX that mandates user authentication.
Peer Trunk: An IP-based SIP trunk that does not require registration. Instead, it relies on IP address authentication for communication between two VoIP systems, such as between your PBX and a VoIP provider or another PBX.
- Note: The configuration settings vary depending on the selected trunk type.
User trunk

Active: Enable this option to activate the trunk for use in all applicable routes. This setting is enabled by default.
Name: Assign a descriptive name to easily identify the trunk.
Auth Type: Select User to configure a registration-based trunk.
Username: Enter the username provided by the VoIP provider for trunk registration.
Password: Enter the corresponding password for trunk registration.
SIP Server: Specify the domain name or IP address of the VoIP service provider.
Port Number: Enter the port number used by the provider.
Caller ID: Define the Caller ID that will be displayed for outbound calls placed through this trunk.
Record Calls: Enable this option to allow recording of both inbound and outbound calls through this trunk.
DTMF Mode: Set the method for transmitting DTMF tones. Auto mode is selected by default.
Transport Protocol: Choose the transport protocol (UDP, TCP, TLS, etc.). By default, UDP is selected, but this can be modified via the Transport settings.
Peer trunk

Active: Enable this option to activate the trunk for use in all applicable routes. This setting is enabled by default.
Name: Assign a descriptive name to easily identify the trunk.
Auth Type: Select Peer to configure an IP-based trunk.
SIP Server: Specify the domain name or IP address of the VoIP service provider.
Port Number: Enter the port number used by the provider. The default SIP port is 5060.
Caller ID: Define the Caller ID that will be displayed for outbound calls placed through this trunk.
Record Calls: Enable this option to allow recording of both inbound and outbound calls through this trunk.
DTMF Mode: Set the method for transmitting DTMF tones. Auto mode is selected by default.
Transport Protocol: Choose the transport protocol (UDP, TCP, TLS, etc.). By default, UDP is selected, but this can be modified via the Transport settings.