I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy.  I
get no ringback audio if call a phone in this direction:

(PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio)

Debugging it, I see that SIP 180 ringing is sent from the phone to
Kamailio, and then relayed properly to Freeswitch, which does recognize
the ring, but because there is no RTP audio associated with it, has
nothing to send into freetdm.

So my question is how do I turn SIP 180 into ringback audio generated on
the PSTN gateway itself?


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