asterisk users - Aug 2007 - Detecting DTMF Tones from Muted app_meetme Participants

Hi, folks.

I have a problem using Asterisk 1.2. I create conferences using
app_meetme and Zap channels, and for every participant I run the script
defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF
tones. As the docs tell me, when using the AGI background script one
loses the ability to control the meetme conference via the command line
so for muting conference participants I had to create a small workaround
(letting the user leave the conference by letting his AGI background
script finish) and then rejoin him using the 'm' flag so that he can
keep listening but is not allowed to talk.

Now, while the user is muted, I can not read any DTMF tones in the AGI
background script. "Okay," you might say, "the user is muted. It
makes
sense that his DTMF tones do not come through." But I say, "the
muting,
i.e. the nullification of any incoming audio is done somewhere within
Asterisk, why is the DTMF tone detection not done _before_ the incoming
audio is thrown away?"

So, basically, my question is, how can I have a muted participant in a
meetme conference that can still control his listening experience via
DTMF tones? And I'm not afraid to mess around in the code - somehow,
somewhere (at least in my imagination) there have to be two lines, the
one reading "remove_audio();" and the other one reading
"detect_dtmf_tones();" which should be rearranged... :)

If anyone has a hint on where to start looking in the moloch that is
Asterisk, please give me a hint. Or if you know in what other way I can
achieve my goal, please give me a hint. I will be grateful until
eternity (or next wednesday, whichever comes first).


Thanks in advance,

	David
-- 
INA Service GmbH
Papenreye 63
22453 Hamburg
Germany

Mail: info at ina-germany.de
Phone: +49 (0)40 557 07-07
Fax: +49 (0)40 557 07-100

Gesch?ftsf?hrer: Ines M. Hoerner

Amtsgericht Hamburg
HRB 96470
Ust-IdNr. DE248754961

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On 8/17/07, David Roden <dr at ina-germany.de>
wrote:
> Hi, folks.
>
> I have a problem using Asterisk 1.2. I create conferences using
> app_meetme and Zap channels, and for every participant I run the script
> defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF
> tones. As the docs tell me, when using the AGI background script one
> loses the ability to control the meetme conference via the command line
> so for muting conference participants I had to create a small workaround
> (letting the user leave the conference by letting his AGI background
> script finish) and then rejoin him using the 'm' flag so that he
can
> keep listening but is not allowed to talk.
>
> Now, while the user is muted, I can not read any DTMF tones in the AGI
> background script. "Okay," you might say, "the user is
muted. It makes
> sense that his DTMF tones do not come through." But I say, "the
muting,
> i.e. the nullification of any incoming audio is done somewhere within
> Asterisk, why is the DTMF tone detection not done _before_ the incoming
> audio is thrown away?"
>
> So, basically, my question is, how can I have a muted participant in a
> meetme conference that can still control his listening experience via
> DTMF tones? And I'm not afraid to mess around in the code - somehow,
> somewhere (at least in my imagination) there have to be two lines, the
> one reading "remove_audio();" and the other one reading
> "detect_dtmf_tones();" which should be rearranged... :)
Oh, if only it were that simple. There are many places within the
app_meetme.c code that parse through DTMF information, and it depends
on the type of channel as to whether it is parsed at all, or how it is
passed along.

I've done quite a bit of messing around with meetme, and most recently
I put up a patch that would do better DTMF passthru between VOIP and
Zap channels in meetme sessions within Asterisk 1.2.X (a backport of
the feature in Asterisk 1.4).
http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch

Of course I must mention that the Asterisk 1.2 branch is in
security-patch-only mode and it is encouraged that you use Asterisk
1.4 to help fix the bugs that are in it and experience some of the
feature enhancements that it has.

MATT---