I am trying to diagnose a SIP issue between our carrier and our network. The carrier has a
CARRIER_IP and a different CARRIER_MEDIA_IP, and it submits an INVITE packet to
MY_PUBLIC_IP using MY_DID. The firewall at our public IP (where the attached traffic
sample was taken) redirects it to a particular Kamailio server at 192.168.10.10 inside the
LAN, and it in turn routes it to the Asterisk instance in localhost. The issue is that the
INVITE is received, then the SIP/2.0 100 Trying and then SIP/2.0 200 OK
are routed back (or so I think), and then the expected ACK from is never received, even
though the caller already hears the media from the Asterisk IVR. After a timeout, our
Asterisk closes the call, in the middle of the conversation.
I am trying to explain the situation to our carrier, but I want to rule out possible
misconfigurations on our side. Are there common misconfigurations that produce the
symptoms described here? Are there any issues evident from the attached traffic?