asterisk users - Feb 2006 - DTMF Mode supported by VoiceMail Application

Hi,

I would like to use Asterisk as VoiceMail system ...
the only issue I have is with DTMF recognition.

Which mode should I force into sip.conf ( general, only for peer ? )
so that the Voicemail application is understanding password from users ...
inband : works, but has some glitch ... not always good ... don't know why.
rfc2833 : doesn't seem to work ..
info : said to be not working .... ( cf
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode )

Did anyone succeed that ?

Thanks a lot !

JMS
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"Jean-Marc Salsa" <jsalsa@gmail.com> writes:
> Which mode should I force into sip.conf ( general, only for peer ? )
> so that the Voicemail application is understanding password from users ...
This depends on what your users are using. If you are using a
Grandstream device you can configure in its administration interface
which dtmf mode the telefone should use. If your IP phone is
configured to use rfc2833 for example then you would write
dtmfmode=rfc2833 in your sip.conf. If all users use the same
dtmfmode it should be ok to write this to the general section. 

Fabian M?ller

Thanks,

But, I do not have phones connected to Asterisk ...
but only one peer : my softswitch ...
So call flow is Phone -> Softswitch -> Asterisk -> Voicemail

I can force the link Sofswitch -> Asterisk ( Codec and DMTF Mode )
Codec is PCMx ...
but as i said inband config is not working all the time !

Let me know if you think something else ...

JMS


On 2/22/06, Fabian M?ller <fabian_mueller@open-tk.de>
wrote:
>
> "Jean-Marc Salsa" <jsalsa@gmail.com> writes:
>
> > Which mode should I force into sip.conf ( general, only for peer ? )
> > so that the Voicemail application is understanding password from users
> ...
>
> This depends on what your users are using. If you are using a
> Grandstream device you can configure in its administration interface
> which dtmf mode the telefone should use. If your IP phone is
> configured to use rfc2833 for example then you would write
> dtmfmode=rfc2833 in your sip.conf. If all users use the same
> dtmfmode it should be ok to write this to the general section.
>
> Fabian M?ller
> _______________________________________________
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On Feb 22, 2006, at 3:44 AM, Jean-Marc Salsa wrote:
> Thanks,
> ?
> But, I do not have phones connected to Asterisk ...
> but only one peer : my softswitch ...
> So call flow is Phone -> Softswitch -> Asterisk -> Voicemail
> ?
> I?can force the link Sofswitch -> Asterisk ( Codec and DMTF Mode )
> Codec is PCMx ...
> but as i said inband config is not working all the time !
> ?
> Let me know if you think something else ...
> ?
It can also be totally dependant on gain settings for inband DTMF in my 
very limited experience.