[Asterisk-Users] DTMF Mode supported by VoiceMail Application

[Jean-Marc Salsa]

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 Müller <fabian_mueller at open-tk.de> wrote:
>
> "Jean-Marc Salsa" <jsalsa at 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 Müller
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060222/53b7dc29/attachment.htm