[Asterisk-Users] Passing DTMF

[Olle E. Johansson]

Eric Wieling wrote:
> 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.

...or the problem is, as hinted, that Asterisk sends a short dtmf.
Regardless of what it receives into the sip channel, Asterisk sends
a 250 ms DTMF signal out (if my memory is correct). You can check
in chan_sip.c
The dtmf setting sets what Asterisk sends to that peer/user.

/O