a quick way to have the caller's UA send the reinvite might be to have them place the call on hold and then resume the call. If this works, it's pretty apparent you do indeed have a reinvite issue though. The carrier might be expecting an empty packet for the rtp keepalive. It might be good to ask the carrier what they would prefer in the following: keepalive interval, method for sip keepalive, method for rtp keepalive and compare that with your settings (as well as a trace).
Our current assumption is that the 1800 seconds is expiring, and we are attempting to initiate the re-invite to the carrier, and the routing is not executing properly, perhaps?
Has anyone experienced this problem? Is there a way to override the request to set the session_expire, or something?
VMK
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