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One account, one extension. Settings › FireTone SIP opens it, and the card at the top of Settings is where you switch the line on and off. The SIP account screen with display name, username, password and server

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The four things you must have

Your SIP password is a credential like any other. Nobody should ask you to read it out, and it is never shown back to you once saved — the field holds dots, and the eye reveals only what you have just typed.

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Transport

Three choices, and they are not equivalent.
  • TLS registers over a secure WebSocket, port 7443 unless you give another. Encrypted, and the right answer wherever the PBX offers it.
  • TCP registers over plain SIP on 5060. Works, and is what most FreeSWITCH installations hand out first.
  • UDP also uses 5060, and cannot carry a FireTone call. The app warns you in red when you pick it. The registration will succeed and then every call will fail, which is a worse outcome than refusing outright.

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Codecs

Transport chooser and the codec list with Opus and G.729 The list comes from the media stack in the app, not from a table written here, so it shows exactly what this build can negotiate. Drag to reorder — the order is the preference order sent in the offer. More codecs reveals the rest. The defaults are Opus and G.729: the best-sounding one and the cheapest one. Opus is 48 kHz and adapts to the network; G.729 is 8 kHz and uses the least bandwidth of anything on the list, which is why PBX trunks are full of it. At least one codec must stay on. Turn them all off and a call cannot be offered at all — the app will refuse to save, and will refuse to join an intercom channel with a message saying so. Low bandwidth roughly halves what a call uses by running Opus narrow and capping the send rate. It needs Opus, and switches itself off if you turn Opus off.

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Advanced

Local port range sets the source ports the phone binds for SIP over UDP and TCP. The default 16000–32000 suits almost everybody; change it only if your network team has asked you to.

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Proving it worked

Test registration at the bottom of the screen registers once and reports back — Test registration succeeded · 240 ms, or the reason it did not. The card at the top of Settings then reads registered, with the transport, the codecs in force and how long the line has been up. If it does not, the SIP log has the reason in full.