asterisk users - Dec 2008 - DTMF pass-through question

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
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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---

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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