On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote: > 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) > > That's for making outbound calls.
This means that you have 2 stage dialing, 9 gives you an outside dial tone. Won't it work with single stage? _9.,1,Dial(${DIALOUTPSTN}/${EXTEN:1}) > Once you've done this, you can direct incoming calls to a context like this: > exten => 50,1,Goto(MainMenu,s,1) > > You should enter 50 to "Forward to VoIP" box at the bottom of HT488 config > page also. (Choose an extension as you like instead of 50) Problem with this is no CallerID it'll always be 50. -- Dave Cotton <[EMAIL PROTECTED]> _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users