Can anyone explain why this is happening?
 
I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted
 
----->------------->--------------------audio in this direction ------>-------------------->-------------------->
[test server that plays .wav file then DTMF digits] -----PSTN-------[Asterisk]-----SIP----[Cisco 7960]
----<------------------------------------call setup in this direction ---------<---------------< ---------------<
 
The Asterisk is set for DTMF=inband , codec g711ulaw

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