[Asterisk-Users] Passing DTMF

[Brian Rathman]

Re: [Asterisk-Users] Passing DTMFI have tried every combination of codec and
dtmfmode. I can hear the dtmf tone on the far end phone, it just appears to
be to short. Is there a way to increase the duration of the tone?
  -----Original Message-----
  From: Eric Wieling [mailto:eric at fnords.org]
  Sent: Tuesday, April 06, 2004 3:10 PM
  To: asterisk-users at lists.digium.com
  Subject: Re: [Asterisk-Users] Passing DTMF


  On Tue, 2004-04-06 at 12:29, Brian Rathman wrote:
  > Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco
  > AS5300 with * in the media stream. Unfortunately, the only way I can get
the
  > calls to connect is with t or T at the end of the Dial() statement and
then
  > that picks off the dtmf digits. I have tried the canreinvite=yes on both
the
  > phone peer and the gateway peer and I still have to add the T to the
Dial
  > statement to make the call complete. Any suggestions???

  cantrinvite=yes tells asterisk to, if it can, remove itself from the
  media stream.  T and t and r and many other Dial options tells Asterisk
  to stay in the media stream so it can listen to the DTMF.  None of this
  has ANYTHING to do with passing DTMF between the two endpoints (except
  of course passing # for t or T).  If you cannot pass DTMF between the
  two endpoints then something ELSE is wrong.  Maybe you are trying to use
  inband DTMF with a compressed codec.  Inband DTMF will only work with
  ulaw or alaw codecs.

  --Eric
  --
  Useful Asterisk Docs (BOOKMARK THEM!):
  http://www.digium.com/index.php?menu=documentation (look at the
  "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and
  http://www.fnords.org/~eric/asterisk/ (my site) and
  http://asteriskdocs.org/

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