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 <[EMAIL PROTECTED]> wrote:
"Jean-Marc Salsa" <[EMAIL PROTECTED]> 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
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