prefixkit Ask an engineerAsk Support RU

Error reference · WireGuard

WireGuard: Name or service not known: `vpn.example.com:51820'

Name or service not known: `vpn.example.com:51820' / Configuration parsing error

What it means

wg-quick could not resolve the Endpoint host name when bringing the tunnel up.

Why it happens

  • DNS is not available yet at boot (the tunnel starts before the network).
  • A typo in the host name, or the DNS record does not exist.
  • DNS works only through the tunnel that is not up yet.

How to fix it

  1. Check the name: dig vpn.example.com.
  2. Start the tunnel after the network: systemctl enable wg-quick@wg0 with After=network-online.target.
  3. Use the server IP in Endpoint if DNS is unreliable.

Useful tools

DNS lookup · WireGuard config generator

Other WireGuard errors

More tools: voip & telephony

AI voice agent cost newRetell, ElevenLabs, Vapi, Bland vs human agents.AI agent ↔ PBX over SIP newAsterisk and FreeSWITCH configs for AI agents.VoIP pcap analyzerCalls, voice quality and why they dropped.SIP log analyzerLadder, SDP, NAT, DTMF and STIR/SHAKEN from a log.TURN/STUN server test newRelay over UDP, TCP, TLS; coturn credentials.RTP stats analyzer newchannelstats, RTPAUDIOQOS, FreeSWITCH: loss, jitter, MOS.NAT type test newSymmetric NAT, UDP and STUN: why audio is one-way.SDP offer/answer analyzer newAgreed codec, DTMF, NAT, hold, one-way audio.CDR analyzer newASR, ACD, NER, causes and fraud signs from CSV.SIP attack log analyzer newBrute-force IPs from Asterisk/FreeSWITCH logs, fail2ban.SIP server checkSRV, TCP/TLS and a SIP OPTIONS ping.IP phone provisioning newYealink, Grandstream, Fanvil config plus DHCP option 66.SIP provider settings newZadarma, Twilio, Telnyx, DIDWW and more: ready configs.Error reference newAsterisk, FreeSWITCH, MikroTik, WireGuard errors explained.SIP trunk generator newAsterisk pjsip.conf, FreeSWITCH gateway, Kamailio.Dialplan pattern tester newWhich Asterisk or FreeSWITCH rule a number hits.Asterisk ⇄ FreeSWITCH dialplan newextensions.conf to XML and back.IVR builder newVoice menu and hours for Asterisk and FreeSWITCH.PBX audio converter newPrompts to 8 kHz WAV, A-law, µ-law; TTS.STIR/SHAKEN decoder newIdentity header: attestation, numbers, checks.VoIP diagnostics newNAT, loss, jitter, bufferbloat and your SIP server in one click.VoIP quality test (UDP) newLoss, jitter, MOS and bufferbloat like a real call.DSCP and QoS newEF, CS3 ↔ ToS ↔ CoS, voice marking configs.MOS calculatorVoice quality from delay, jitter and loss.VoIP bandwidthBandwidth per call with all headers and VPN.Erlang B & CTrunk channels and call-centre agents needed.SIP response codesWhat every code means and what to check.Q.850 causesCall release causes and their SIP mapping.
Idea?