> ## Documentation Index
> Fetch the complete documentation index at: https://docs.averohq.sk/llms.txt
> Use this file to discover all available pages before exploring further.

# Telephony (SIP)

> Connect any SIP phone line or trunk to Avero.

## How telephony works in Avero

Avero is a **bring-your-own SIP** application. It does not provide phone numbers or carrier services — you connect your existing SIP line, PBX extension, or SIP trunk, and Avero handles all calling through it.

This means:

* You keep your existing phone numbers and carrier contracts
* Calls are billed by your carrier, not Avero
* No porting, no number acquisition, no hidden per-minute fees from Avero

## Supported setups

| Setup | Example providers |
| - | - |
| Hosted PBX / UCaaS extension | 3CX, FreePBX, Asterisk, FusionPBX |
| SIP trunk (direct carrier) | Twilio, Vonage, T-Mobile SIP, Slovak Telekom |
| VoIP provider account | VoIP.ms, SipGate, Zadarma |

Any SIP provider that supports standard SIP registration (RFC 3261) works.

## Connecting a SIP line

1. Go to **Settings → Telephony**.
2. Click **Add SIP account**.
3. Enter your credentials:
   * **SIP server** — host or IP of your registrar (e.g. `sip.yourprovider.com`)
   * **Username** — your SIP account username or extension number
   * **Password** — SIP password (not your web portal password)
   * **Port** — default 5060; use 5061 for TLS
4. Click **Connect** — Avero registers and shows **Connected** when ready.

<Tip>
  Get these credentials from your PBX admin or carrier's SIP trunk settings panel, not from their web login.
</Tip>

## Network requirements

Avero uses SIP over UDP (port 5060) and RTP for audio (ports 10000–20000 by default). If you're behind a corporate firewall or VPN:

* Allow outbound UDP 5060 for SIP signaling
* Allow outbound UDP 10000–20000 for RTP audio
* If your firewall performs strict NAT, enable **STUN** in the SIP settings (`stun.averohq.sk:3478`)

## Caller ID (DID numbers)

If your SIP trunk includes multiple DID numbers, you can assign them in **Settings → Telephony → Numbers**. Avero will show which number is used for outgoing calls.

For **local presence** (matching caller ID to contact area code), you need DIDs in the regions you call. Configure the matching rules under **Settings → Telephony → Local presence**.

## Call recording

Avero captures audio locally on your device for AI transcription. It does not upload raw audio to any server — only the text transcript is stored. If your carrier or PBX also records calls server-side, those recordings are separate and governed by your carrier's terms.

<Note>
  Always comply with local call recording consent laws. In most EU countries (including Slovakia), you must inform the other party that the call is being recorded.
</Note>

## Troubleshooting

**Not registering (401 Unauthorized)** — double-check the SIP username and password. Make sure you're using the SIP credentials, not the web portal login.

**One-way audio** — the RTP ports may be blocked by your firewall. Enable STUN or ask your IT team to open UDP 10000–20000 outbound.

**Calls dropping after 30 seconds** — often a NAT keepalive issue. Enable **SIP OPTIONS ping** in the advanced settings to keep the registration alive.

**Echo on calls** — enable **Echo cancellation** in **Settings → Telephony → Audio**.


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