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

[jonathan augenstine]

Matt,

Asterisk version == 1.4.22
dtmfmode == info
calls are bridged through Asterisk (canreinvite=no)

Jonathan

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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