[asterisk-speech-rec] Sound not being recognized ~ multiple asterisk servers? SIP issue?

[Chris Rider]

I have a development box running Asterisk, behind a company network also 
running the same.

The company PBX is properly configured (as far as can be tested) to 
forward any calls made to an arbitrary extension (ext. 80) to my dev 
box, whose Asterisk then handles the call.

I've also setup a softphone on my testing laptop (ext. 1000) to connect 
directly to my devbox.

I am trying to get Vestec speech recognition up and running; and have 
suceeded 50% (works on my laptop, but not from my SIP phone). BTW, using 
standard vestec digit4 demo.

At first glance, it appears like voice/sound data is not coming in to my 
dev box (A.K.A. Asterisk/Vestec server) ~ even though it handles the 
call correctly. I hear the vestec demo playback, prompts, see the call 
handled in the CLI, etc. It just won't hear my speech or dial tones. 
However, my laptop with the "direct" connection works perfectly.

Anyone know of any SIP / multiple PBX issues with getting Vestec to work?

I can post sip.conf and extensions.conf if needed; but doesn't seem to 
be the problems.

-CR