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Asterisk and the one that doesn't work returns 100 trying followed by 183
session progress.

It is my understanding that 180 ringing causes ringback to be generated by
the callee, while 183 means that the caller has early media and will send
ringback through RTP.

Anyone have any idea why I wouldn't get ringback in this case?

Should Asterisk be passing through the early media to the first caller
even though the second caller has not answered?

I am not using the "r" option to the Dial command. I have tried it both
on and off and get no ringback in either case. I have also tried variations
of the progressinband setting.

I have listened to the RTP going from the SIP server to Asterisk and I can
hear the ringing in it. It seems like Asterisk isn't sending any audio to
the first caller until both parties answer.

Thanks,
Matthew Boedicker