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[asterisk at voipbusine...]

All;
    I have a problem that I’ve been working on for a while now, but I’m stuck and can’t see what the solution is. I have an Asterisk 1.11 server on a public IP address and have two phones registered from behind a NAT. I can send a page to/from each phone without a problem. My problem is that if I play an audio file over a page, the page disconnects after a few seconds ( seven seconds to be exact ).
I’m playing the audio file like so:
exten => s,n,Page(${AVAILCHANS},A(demo-congrats,q)

In the CLI I’m seeing this:
[2015-03-27 11:40:26.360] Got  RTP packet from    X.X.X.X:2256 (type 00, seq 021523, ts 1374867997, len 000160)
[2015-03-27 11:40:26.362] Sent RTP packet to      X.X.X.X:2256 (type 00, seq 050875, ts 050560, len 000160)
[2015-03-27 11:40:26.363] WARNING[11325][C-0000002d]: pbx.c:6709 __ast_pbx_run: Timeout, but no rule 't' or 'e' in context 'scheduledpages'

Where X.X.X.X is the outside IP address where the phones are coming from. I’m seeing the GotóSent messages several hundred times while the audio is playing. Like I said, simply paging an extension with a human voice works just fine. Any insight at all would be greatly appreciated.
Thanks much;
John