SIP Error Codes & Troubleshooting Guide

this table lists sip response codes you may encounter , what they usually mean in real deployments, where the issue most likely lies , and what to check next 1\) provisional responses (1xx – informational) sip code meaning what it means what to check 100 trying sip request received and being processed normal behavior 180 ringing destination endpoint is ringing normal behavior 183 session progress early media (ivr / announcements) if audio missing → rtp ports 2\) successful responses (2xx) sip code meaning what it means what to check 200 ok call successfully connected if no audio → rtp / nat 3\) client errors (4xx – most common) sip code meaning likely root cause what to check 400 bad request malformed sip headers or sdp sip logs for invalid headers 401 unauthorized authentication required credentials (if applicable) 403 forbidden ip not allowed acl / whitelisted ip 404 not found invalid did or routing phone number mapping 408 request timeout no sip response firewall / reachability 410 gone number inactive or removed did status 415 unsupported media codec not supported ensure g 711 a law enabled 480 temporarily unavailable endpoint unreachable pbx availability 484 address incomplete invalid number format e 164 format 486 busy here destination busy normal behavior 488 not acceptable here sdp / codec mismatch g 711 a law only 4\) server errors (5xx) sip code meaning likely root cause what to check 500 server error internal processing failure retry call 502 bad gateway upstream routing issue retry 503 service unavailable destination unreachable network / routing 504 gateway timeout no upstream response firewall / latency 580 precondition failure media negotiation failed rtp / nat 5\) global failures (6xx) sip code meaning likely root cause what to check 600 busy everywhere all endpoints busy retry 603 decline call rejected by endpoint pbx dialplan 604 does not exist anywhere invalid number did correctness 606 not acceptable media policy failure codec / sdp 6\) media (audio) issues – no sip error shown these are the most common issues and do not always show sip error codes symptom likely cause what to check no audio both ways rtp blocked udp 10000–40000 open one way audio nat / firewall public ip in sdp audio drops after answer rtp timeout firewall idle timeout ivr audio but no agent audio asymmetric rtp symmetric rtp enabled 7\) codec requirements (india) item requirement supported codec g 711 a law other codecs not supported common failure 415 / 488 errors ensure g 711 a law is enabled no other codec is forced ahead of it transcoding is disabled (recommended) 8\) api level errors (exotel) api error code http status meaning what it indicates common endpoints what the customer should do 1000 404 not found resource does not exist post /trunks/{trunk sid}/phone numbers post /trunks/{trunk sid}/whitelisted ips post /trunks/{trunk sid}/destination uris put /trunks/{trunk sid}/phone numbers/{id} verify trunk sid, phone number id, or api path 1001 400 mandatory parameter missing required field not sent in request post /trunks post /phone numbers put /phone numbers/{id} ensure all mandatory fields are present 1002 400 invalid parameter field value is invalid or not allowed post /trunks post /destination uris post /whitelisted ips validate parameter format (ip, fqdn, e 164 number) 1007 400 invalid request body json is malformed or not parsable all post / put apis fix json syntax, remove trailing commas, ensure valid json 1008 409 duplicate resource resource already exists post /trunks post /phone numbers post /whitelisted ips safe to ignore if config already exists 1011 415 unsupported content type incorrect content type header all post / put apis use content type application/json 9\) before contacting exotel support please verify sip signaling port (443 / 5070) reachable udp 10000–40000 open correct exotel ips allowed correct trunk domain used correct mode (pstn or flow) correct destination uri configured g 711 a law enabled 10\) when to contact exotel support raise a ticket at support exotel com(hello\@exotel com) only after verification please share trunk sid callsid call direction (inbound / outbound) sip response code (if any) timestamp (with timezone) pbx sip logs or pcap (if available)