[Asterisk-Users] sip -> h323 -> ptsn

[Brian West]

thats all we use right now

On Wed, 30 Jul 2003, Eric Wieling wrote:

> That only works if you are using the G711 (ulaw/alaw) codecs.  Other
> codecs distort inband DTMF.
>
> On Wed, 2003-07-30 at 15:26, Patrick wrote:
> > I have the same setup, and in the sip.conf file I set the dtmfmode=inband
> > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
> >
> >
> > On Wed, 30 Jul 2003, Brian West wrote:
> >
> > > I have this setup:
> > >
> > > Sip Phones -> Asterisk -> h323 gateway -> ptsn
> > >
> > > Sip phones are setup for out of band dtmf
> > >
> > > but the h323 gateway is inband.  Is their a way to pass the digits from
> > > the sip phones to the ptsn via the h323 gateway?
> > >
> > > bkw
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> > >
> >
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