On Wed, 2005-08-31 at 09:54 -0300, Keith Yoder wrote: > Soner Tari escreveu: > > > I use HT488, and I can make and receive FXO calls. It's actually quite > > simple, you create a SIP acount in sip.conf. On the FXO section of > > HT488 web admin page you enter these registration values. When you > > reboot the HT488 you should see it registering on Asterisk CLI. > > > > What's left is a dialplan line in extensions.conf like this: > > exten => 9,1,Dial(SIP/<sip acount name>,10) > > > I've tried your example shown here. When I dial 9 I get dial tone from > the HT488 but I when I try to dial, nothing happens (i.e. I keep hearing > dial tone even though I'm dialing). Any ideas?
What are your DTMF settings? I had all sorts of weird problems with a differant manufacturers ATA because of this. -- Dave Cotton <[EMAIL PROTECTED]> _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
