Build custom call flow for ATA in Kamailio

We need someone to build a custom call flow for an ATA connected on a SIP platform with a kamailio proxy load balancing 2 freeswitch servers. There is currently no audio on incoming calls on the ATA. Outgoing calls are working fine. Incoming calls using a softphone are working fine. The ATA is working on another platform using Kamailio load balancing two Asterisk servers. This might be due to some ATA specific way of handling the SIP messages and hence needs a workaround. We need someone with extensive experience of VoIP systems to setup this new call flow on kamailio. Ongoing consulting work possibility.