> For the complete documentation index, see [llms.txt](https://docs.limephone.io/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://docs.limephone.io/calls-and-messages/sip-dialer.md).

# SIP dialer

Ring your calls on a SIP endpoint — a desk phone, PBX extension, or softphone app — alongside or instead of the browser.

## Enabling SIP ringing

1. Sign in at [dashboard.limephone.io](https://dashboard.limephone.io) and open **Settings → Your account → Call Settings**.
2. Turn on **Ring via SIP**.
3. Enter your **SIP URI** — the address of your endpoint, in the form `sip:user@domain.com` (for example `sip:john@sip.example.com`).
4. Save. Incoming calls that route to you now ring that SIP endpoint, simultaneously with any other devices you have enabled (**Ring in Browser**, **External Numbers**).

## Getting a SIP URI

Your SIP URI comes from your SIP provider, PBX, or softphone:

* **Softphone apps** (e.g. on desktop or mobile) — register the app with a SIP account first; the account's address is your URI.
* **PBX / VoIP system** — ask your administrator for the extension's external SIP address.

LimePhone sends the call to that URI; your endpoint must be registered and reachable from the public internet to receive it.

## Troubleshooting

* **SIP device never rings** — confirm **Ring via SIP** is on and the URI has no typos (it must start with `sip:`). Then confirm the endpoint is registered with its provider — an unregistered softphone silently misses calls.
* **Rings but no audio** — usually NAT/firewall on your side; check your SIP device or router's SIP/RTP settings, and see [Call quality](https://docs.limephone.io/troubleshooting/call-quality).
* **Still stuck** — open a ticket with [Support](https://help.limephone.io/support/tickets/new) including your SIP provider and the time of a test call.