AdminGuide:BasicConcepts:Outbound lines:VoIP Termination

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For every VoIP domain, you can set up any combination of VoIP terminations and trunks.

For outbound lines, it is possible to use each as an independent line. For inbound lines, all calls from that domain will be handled by the "VoIP domain" entity, which contains all the routing rules related to the numbering of the trunks and terminations that belong to it.

The following table lists the configurable parameters for a VoIP termination.

Parameter	Description	Value
-
Enabled	Lets you disable a VoIP termination without losing its configuration.	On / Off
Name	Mnemonic name assigned to the VoIP termination.	Alphanumeric
SIP settings
VoIP domain	Select the hostname of the VoIP domain to which you wish to associate the termination.	Hostname
Source user	Name of the source	Alphanumeric
Username	Username with which the trunk is accessed.	Alphanumeric
Secret		Alphanumeric
Transport settings
Transport type	Lets you choose between SIP with UDP/TCP/TLS transport or SIP with WebSocket/Secure WebSocket transport for this domain.	SIP / WebRTC
Preferred transport	Drop-down menu that lets you choose the preferred transport among those enabled. Selecting a disabled transport will automatically enable it.	UDP / TCP / TLS / WS / WSS
Enable UDP transport	Enable unencrypted UDP transport. Only available if "Transport type" is set to SIP.	Yes / No
Enable Web RTC transport	Enable Web RTC transport protocol.	Yes / No
Advanced settings
Simultaneous call limit	Number of simultaneous calls allowed.	Numeric
DTMF mode	Choose how DTMF tones are sent to this gateway, among the modes provided (RFC 2833, SIP Info, in audio). By default, this will be set to the predefined system mode.	RFC 2833 / SIP Info / In audio
COLP sending mode		System default / Disabled / Remote-Part-ID / P-Asserted-Identity
COLP acceptance mode		System default / Disabled / Enabled