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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

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Supported setups

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

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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.
Get these credentials from your PBX admin or carrier’s SIP trunk settings panel, not from their web login.

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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)

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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.

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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.
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.

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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.