asterisk users - Feb 2009 - DTMF not completely muted

Hi all,

When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15
ms
and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
voicemail greetings -- you hear a very short DTMF '#', or sometimes two
blips,
at the end of the recording.

I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards:
a TE420 w/Octasic and pri_net signalling, and a TE220 w/o Octasic using em_wink
signalling. Calls come in using either PRI or E&M depending on Mitel routing
(you don't want to know...).

Either way, the driver is doing software DTMF detection. (Looking through the
DAHDI driver, hardware DTMF detection is always disabled by default.)

The calls are then sent to another Asterisk instance by SIP with RFC2833 for
DTMF. I dumped the RTP going back and forth between the Asterisks -- the RFC2833
DTMF is being sent properly, but the blips are in the audio stream.

Does anyone have suggestions for preventing the blips, or cleaning them up
afterward? I was thinking I could run a script over all voicemail greetings to
find the first DTMF tone and crop the wav file, but it's not so easy -- the
DTMF
blips are way too short to be picked up by spandsp's DTMF detector.

I have some SIP phones and they of course don't have any problems recording
voicemail greetings without blips -- they all use RFC2833 and no real tones are
in the audio stream.

Mike

Michael Smith wrote:
> When the Dahdi driver detects DTMF, it seems it's not muting the first
5-15 ms
> and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded
> voicemail greetings -- you hear a very short DTMF '#', or sometimes
two blips,
> at the end of the recording.
The blip at the beginning is expected; the DTMF detector won't trigger
until it has seen a long enough continuous tone to recognize it, and
then it will begin muting the audio. The only way around this would be
to artificially delay the audio stream by 20-40ms to accommodate the
muting, but most people would not enjoy the additional latency this
would create.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org