Device settings in depth

The device edit page is one long form, grouped into sections. This page walks through the sections an operator actually touches: authentication, NAT/transport, codecs, caller ID, call limits, recording and language.

Open a device from the devices list. The edit page has tabs — General, Callflow, Codecs, CLIs, CID pools, Rules, Voicemail — plus, on the General tab itself, a live PJSIP configuration preview on the right showing exactly what will be written to Asterisk once you save.

Authentication & network

This is how the phone proves who it is, and where Voiplix expects to find it.

Authentication and network section: IP authentication checkbox, Secret (masked), Dynamic host registration checkbox, Port, NAT, Transport. — Authentication & network — Secret is masked here for the screenshot; the real form shows it in the clear to whoever has edit rights.

Field	Meaning
`IP authentication`	Trust the phone by its IP address instead of a username/password. Turn this on only for a device at a fixed, known address.
`Secret`	The SIP password the phone uses to register. Hidden when IP authentication is on — there is nothing to authenticate with a password at that point.
`Dynamic host (registration)`	Checked: the phone registers itself and its address can change (the normal case for a phone on a home or office connection). Unchecked: you type a fixed Host / IP instead.
`Port`	The SIP port to expect the phone on. `0` lets Asterisk use whatever the registration reports.
`NAT`	How to handle a phone behind a router doing network address translation. Leave at the default unless a phone behind NAT has one-way audio or registration trouble.
`Transport`	The SIP transport: `udp` for a normal phone, or a TLS/WebSocket option for an encrypted or browser-based connection.

Warning

A device without IP authentication and without a Secret is unprotected — anyone who guesses or finds the extension can register as that phone. The devices list has a dedicated Weak passwords check for exactly this.

Codecs

Open the Codecs tab. Active (order) lists the codecs this device is allowed to use, in the order they are offered during call setup — the order is the negotiation priority. Move an entry with the arrows, or Remove it back to Available on the right.

Tip

Fewer, well-chosen codecs mean less negotiation and less chance of a mismatch with the far end. G.711 (alaw/ulaw) is the safe, universal choice; g729 uses less bandwidth but needs a license on some Asterisk builds.

Caller ID

The CallerID section decides what number this device presents to the person it calls. It has several modes, chosen with the radio buttons under Number:

Mode	Meaning
`Number:`	Always present this one fixed number.
`Number from DID`	Present the DID that this device is reached through.
`Control by DIDs` / `Control by CIDs`	Look up the presented number from the DIDs or CID rules assigned to this device.
`Control by destination`	Choose the presented number based on where the call is going.
`Random number from number pool` / `Random number pool from CID pools`	Present a different number from a pool each time — useful for spreading outbound traffic across several numbers.
`Unknown`	Present no caller ID at all.
`Copy leg A name to leg B number`	Carry the inbound caller's name through as the outbound number, in transfer/forward scenarios.

Below the mode, Block CallerID at N simultaneous calls hides the number once this device has that many calls in progress, and Emergency CallerID is the number sent instead whenever the dialled number matches the emergency number pool — get this one right for any device that can dial emergency services.

Call limits

On the General tab, Call limit caps how many simultaneous calls this device can have (0 = unlimited). Further down, under Advanced network, Time limit per day (min) and Time limit per month (min) cap total talk time, with a Day of reset for the monthly one — both are 0 = unlimited by default.

Recording & language

The Language field on the General tab sets which language this device hears system prompts, voicemail instructions and announcements in (for example en). The Recordings section, near the bottom of the form, controls automatic call recording for this device.

Debug and Recordings sections: Process SIPCHANINFO, Save the SIP trace automatically, Record calls for this device, Show recordings to the user for this device, Send recording to email, Recordings email, Keep recordings on disk after sending by email. — Debug & Recordings — recording, and who gets to see or receive the recording.

Field	Meaning
`Record calls for this device`	Turn on automatic recording for every call this device makes or receives.
`Show recordings to the user for this device`	Let the owning customer play back this device's recordings from the customer portal.
`Send recording to email` / `Recordings email`	Where to send a copy of each recording. Saved, but not active in this version — sending recordings by email is not connected yet.
`Keep recordings on disk after sending by email`	If email delivery is ever turned on, this decides whether the local copy is kept afterwards.

Check

Save the device, then look at the PJSIP configuration preview on the right of the General tab — it should show the codecs you chose (allow=), the transport and NAT settings, and (once a call is made) match what actually happens on the line.