Generic SIP

For Asterisk/FreePBX, 3CX or any SIP trunk that is not listed. It has no region: Turkish and international numbers can be used.

Values are taken one-to-one from the product’s telephony preset (sip_presets.py). Check any field marked Not verified in the docs in your provider’s panel; it stays editable in the console with the same note.

Provider key

generic_sip

Region

Any (TR and non-TR)

Connection test

SIP REGISTER

Username

Trunk username (may stay empty with IP authorization)

Caller ID (DID)

Caller number (DID), e.g. 0850 123 45 67

Preset values

These values are pre-filled in the console; you can change every one of them.

Field	Default	Status
SIP server	enter your own — domain or IP, optional :port	! Not verified in the docs
Transport	UDP	! Not verified in the docs
Authentication	username / password (Digest)	! Not verified in the docs
Dialled number format	national	! Not verified in the docs
Inbound signalling IPs	— (none published)	! Not verified in the docs
Suggested codecs	PCMA, PCMU	! Not verified in the docs
SIP REFER (transfer)	unknown — use bridge	! Not verified in the docs

Provider notes

Asterisk/FreePBX, 3CX or any other SIP trunk. Define the LiveKit SIP server's IP as a trunk on your PBX; authentication can be username/password or IP authorization.

Connecting

1. In Settings → Integrations → Phone line, pick Generic SIP as the provider. The preset fills in the fields.
2. Enter the SIP username, password (write-only) and caller number (DID); change the server, transport or number format if needed.
3. Press Test connection. Then map your inbound numbers to an assistant. Number mapping →