Markus wrote:
Hi Joshua,
Hola,
My suggestion is to take a step back further.
Just send incoming calls to the Read application and have it store the
received DTMF in a variable. Next step have it output what was received.
Ok, good idea, here are the results of Read() and SayDigits():
<snipped results to make this email manageable>
How are you changing the DTMF for each provider? If you are merely
changing it using dtmfmode in sip.conf this may or may not change how
the provider side sends it. In the case of setting it to rfc2833 it
causes RFC2833 to be negotiated in the SDP. Some equipment MAY change to
using inband if it has not been negotiated.
If that works for all cases then Asterisk is recognizing DTMF fine. This
does *not* mean that the tone will be muted fully as my previous email
mentioned.
I don't see any previous eMail from you on the list and there is nothing
in the archives either. Could you re-send it, please? Maybe the info
that I'm missing is inside that mail. :)
I think I wrote the email in my head, oops.
Essentially when doing conversion of inband DTMF to out of band DTMF it
is possible for some parts of tones to get through unmuted. You have to
strike a balance between detecting the DTMF early enough, not detecting
other stuff as DTMF, and muting it. Some implementations may let some
"leak" through. The only way to completely overcome that is to buffer
enough audio and delay the stream.
You can further test if all cases check out by sending calls to Record
and playing back the audio to yourself. If you hear tones and Asterisk
also recognized the DTMF then it's not fully muted, or the hardware in
question is sending *both* inband and out of band, which is not
supported.
Ok, here are the results of Record() and Playback():
<snipped results again, can be viewed in mailing list archives if anyone
is curious>
If I understand right, all my four DID providers are "broken"?
If the provider is doing the conversion their equipment should mute the
inband DTMF as best it can and you should not hear it.
How much of the tone are you hearing?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
