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

[Patrick]

Which codec are you using?  and which H.323 channel driver?  chan_h323 or 
chan_oh323 ?


On 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
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>