[asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

[Yaron Nachum]

Hi Mathew,
The regular sip stack has 'auto' dtmfmode which behaved as I said - if the
remote replied with telephony event it used RFC2833 otherwise it used
inband.




On Tue, Mar 11, 2014 at 5:43 PM, Matthew Jordan <mjordan at digium.com> wrote:

> On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum <nachum.yaron at gmail.com>
> wrote:
> > Hello,
> > I have installed the latest version 12 that has been released
> (12.1.0.rc3).
> >
> > I have setup default dtmf mode (rfc47..) but when I am calling to a
> endpoint
> > that doesn't support it (no telephony event in the rtpmap) the asterisk
> > responds OK in the signalling but DTMF is not working.
> >
> > Is it a known issue?
> >
>
> I don't think that's an issue at all.
>
> Your configured your endpoint to support RFC 4733 DTMF. However, the
> INVITE request that was received by Asterisk didn't offer support for
> DTMF, so Asterisk can't accept it. It has to accept only what is in
> the offer.
>
> Your configuration can't force the UA to offer what it wants - you can
> only configure Asterisk with what it should support with that UA.
>
> There's really only two possible outcomes here:
> (1) Reject the INVITE request with a 488 (you didn't offer me DTMF!)
> (2) Accept the INVITE request but not have DTMF over RFC 4733.
>
> What you're seeing is option (2), which I think is better than
> rejecting the entire call simply because the thing you are talking to
> doesn't support the DTMF mode you configured it to have.
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
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