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

Eric Wieling eric at fnords.org
Wed Jul 30 14:22:11 MST 2003


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