[Asterisk-Users] Weird DTMF issue

[Lee Goodman]

    Sorry, I miss typed , I meant XLite not XTEN as the softclient I was talking about :)
  ----- Original Message ----- 
  From: Lee Goodman 
  To: asterisk-users at lists.digium.com 
  Sent: Wednesday, August 13, 2003 9:40 AM
  Subject: Re: [Asterisk-Users] Weird DTMF issue


  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 ----- 
    From: Lee Goodman 
    To: asterisk-users at lists.digium.com 
    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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