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
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