Troubleshooting

Start with resource status:

kubectl -n telephony get networkprofile,sipgateway,mediarelay,asteriskpool,sipuser,siptrunk,callscope,dialpolicy,callroute
kubectl -n telephony describe sipgateway main
kubectl -n telephony logs deployment/kubevoip-kubevoip

For SIP routing issues, inspect Kamailio summary logs:

kubectl -n telephony logs deploy/main-sip-gateway -c kamailio \
  | grep kubevoip_sip_event

For one-way audio, check RTPengine Service addresses, firewall rules, and RTP port forwarding. SIP and RTP public ports must match the private ports. RTPengine startup and runtime logs are available from the relay pods:

kubectl -n telephony logs deploy/main-rtpengine-0 -c rtpengine

For deeper call-flow analysis, enable HOMER HEP capture on the affected SIPGateway and search HOMER by SIP Call-ID.