[asterisk-users] [Asterisk-users] DTMF pass-through question

[Matt Florell]

On 12/28/08, jonathan augenstine <jaugenstine at gmail.com> wrote:
> Matt,
>
> Asterisk version == 1.4.22
> dtmfmode == info
> calls are bridged through Asterisk (canreinvite=no)
>
> Jonathan

Have you tried setting dtmfmode to 'inband' for both SIP endpoints?

MATT---

> On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell <astmattf at gmail.com> wrote:
> >
> > On 12/28/08, jonathan augenstine <jaugenstine at gmail.com> wrote:
> > > I am trying to resolve an issue and I believe it is my configuration.
> The
> > > scenario is that I have a SIP detected on the server.  The dial plan
> then
> > > makes a local connection to another part of the dial plan.  The new dial
> > > plan extension then places another SIP call out to a SIP phone.  When
> the
> > > call is accepted there is streamed from the calling SIP phone.  When the
> > > audio is complete a DTMF is transmitted to Asterisk.  The DTMF is
> detected
> > > by Asterisk but it does not get passed through to the other SIP phone.
> I
> > > would like the DTMF to pass-through to the other SIP phone.  Is this a
> > > configuration issue?  Or do I need to handle this on the dial plan
> level?
> > >
> > > Jonathan
> >
> > Asterisk version?
> >
> > What are both dtmfmodes set to for each SIP endpoint?
> >
> > Are the calls natively bridged or bridged through Asterisk?
> >
> > MATT---
> >
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