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[asterisk at voipbusine...]

All;
I’m running Asterisk 11.6-cert9 and am trying to play a pre-recorded audio file to extensions using the Page() command. The dial plan looks like this:
exten => s,n,Page(${AVAILCHANS},A(${AUDIOMSG})) and the paging by itself works great. However, when I try it with the audio file, it starts to play correctly, then abruptly hangs up after 6 or 7 seconds. When I turn debug on, this is what I see:

[2015-03-19 15:46:38.292] Sent RTP packet to X.X.X.X:1049 (type 00, seq 037511, ts 061440, len 000160)
[2015-03-19 15:46:38.299] Got RTP packet from X.X.X.X:1049 (type 00, seq 036666, ts 4175299232, len 000160)
[2015-03-19 15:46:38.312] Sent RTP packet to X.X.X.X:1049 (type 00, seq 037512, ts 061600, len 000160)
[2015-03-19 15:46:38.316] Got RTP packet from X.X.X.X:1049 (type 00, seq 036667, ts 4175299392, len 000160)
[2015-03-19 15:46:38.323] Got RTP packet from X.X.X.X:1049 (type 00, seq 036668, ts 4175299552, len 000160)
[2015-03-19 15:46:38.329] WARNING[25939][C-00000000]: pbx.c:6709 __ast_pbx_run: Timeout, but no rule 't' or 'e' in context 'scheduledpages'

I’m thinking that this could be NAT related. My Asterisk server has a public IP address, but my extensions are behind a NAT if that helps any. My extension configs have “nat=force_rport,comedia”. I could really use some insight with this and would be very grateful for any help at all.
Regards;
John