[Asterisk-Users] SIP <-> PSTN DTMF

[goldhorse]

Hi,

I use a Handytone 486. While you have to configure dtmfmode=rfc2833 in 
asterisk,
it will not work if you do not set the dtmf mode to "SIP info" in the 
ATA itself.
So you might try different combinations fo dtmf modes in both asterisk 
and
the phone you are using until you get the correct one.

Hope this helps

Daniel

On 2005/03/19, at 14:27, John Goerzen wrote:

> Hi,
>
> I have a SIP phone connecting to my asterisk server, using
> dtmfmode=rfc2833.  When calling from the SIP phone to internal asterisk
> services, such as voicemail, it works fine.
>
> But when I call out to the PSTN, from the SIP phone, via my X100P, the
> call will be connected fine.  After that, though, any numbers I dial on
> the SIP phone are lost.  I hear them on the phone, but I don't hear 
> them
> on the remote end of the PSTN connection.
>
> I know that rfc2833 is correct for the SIP phone since it is working
> fine with internal asterisk services.
>
> I have tried the wiki, searching the list, and google.  No luck.  Ideas
> would be welcome!
>
> -- John
>
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