Ok, I think I figured out the problem
 
If both the phone and asterisk are using RFC2833, I get this DTMF problem. But, If I set both the phone and Asterisk to DTMF=inband, the DTMF tones sound much better. (I verified the DTMF status of the call by doing "sip show channel ?" in the CLI)
 
Is there a known problem with RFC2833 in Asterisk? (Note, this would only happen when Asterisk acts as a SIP endpoint in the Call, like when a SIP phone calls out through the FXO port not when the call is just passing through the server).
 
There is also a thread about how the XTEN softphone seems to have a DTMF problem. XTEN says they fixed it, but people (including myself) still see the problem. I wonder if there is a bug in Asterisk support of RFC2833? One of the XTEN users says that one way the problem shows up is if you send a string of DTMF's that are the same number (1001 fails , but 1234 works). I have seen the same issue on a Cisco phone ascessing VM on an Asterisk server.
 
Lee Goodman
----- Original Message -----
Sent: Tuesday, August 12, 2003 4:32 PM
Subject: [Asterisk-Users] Weird DTMF issue

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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