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 > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
