Asterisk Auto Scaling

We use a SIP Trunk vendor which does not allow adding IPs to their Allow List programatically. As such, we are constrained to use one IP publicly and redirect it individually to each asterisk servers within a network.

Our solution is to “redirect” all SIP registrations from a dispatcher (publicly exposed IP) to asterisk servers via round robin. This setup calls the number number as intended but theres no audio on both ends. Conceptually I understand why this is possibly happening,

Normal Setup: (working)
1.) Softphone -> Asterisk Server
2.) Asterisk Server -> SIP Trunk Vendor
3.) (upon answer) SIP Trunk Vendor -> Asterisk Server -> Softphone (reversed steps 2 & 1)

Auto Scaling Setup:
1.) Softphone -> Dispatcher
2.) Dispatcher -> Asterisk Server -> NAT Gateway -> SIP Trunk Vendor

However since NAT Gateway is outbound only, reversing step 2 is not possible like in Normal Setup. Ideal setup should be:
3.) SIP Trunk Vendor => Dispatcher -> Same Asterisk Server -> NAT Gateway -> Softphone

Any ideas on how this can be possible? Or if openSIPS/kamailio should be used for this. Thanks!

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