prefixkit Ask an engineerAsk Support RU

Error reference · Asterisk

Asterisk: Retransmission timeout reached on transmission 1a2b3c@203.0.113.5 for seqno 102 (Critical…

Retransmission timeout reached on transmission 1a2b3c@203.0.113.5 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

What it means

Asterisk resent a SIP request for 32 seconds and never got a reply: the other side or the path is gone.

Why it happens

  • The remote device went offline or changed IP.
  • NAT or a firewall drops the packets on the way back.
  • Large SIP packets fragmented over UDP and lost (big SDP, many codecs).

How to fix it

  1. Check reachability of the peer (ping, SIP check).
  2. For NAT: correct external_signaling_address and local_net; keepalives/qualify.
  3. Move to TCP for peers that send large messages, or trim codec lists.

Useful tools

SIP server check · MTU calculator

Other Asterisk 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?