[asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

[Markus]

Am 28.09.2012 17:33, schrieb Joshua Colp:
>> 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.

Yes, only via dtmfmode in sip.conf. I have no control over the provider 
side, of course. :)  Or are there other options?


> 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.

Ah! So because I'm calling from PSTN which is "naturally" inband (I 
guess?) my DID provider (or rather some provider in front of him) 
converts to out of band (RFC2833), but the conversion cannot be fully 
accurate?


>> 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?

About the tone itself: it's always different for each DTMF 
method/provider I would say. Sometimes it's just a really short "glub" 
(no idea how to describe that better), and sometimes it's a full-length 
"beeep". But in any case except for provider 4 inband + INFO I'm hearing 
all of it (123), so it's anything from glub-glub-glub to 
beeep-beeep-beeep. :-)

Right now, after what you wrote, I'm left with the feeling that RFC2833 
is pretty much, well, hmm... useless in a PSTN-VoIP scenario? It works 
fine when using X-Lite, I suppose because there is no conversion 
involved and therefore the tones can get completely muted.

I'm sad now. Is there nothing I can do to remove DTMF tones from my 
conferences? :-(

Thank you!
Markus