On Sunday 09 May 2004 02:32 pm, bob mc wrote: > Hi, > > I'm trying to put together a simple gateway > configuration involving Asterisk. I have a machine > with 2 Digium X100P FXO cards installed and the > Asterisk Software, and I have 2 Sip Phones defined. > > What I want to achieve is, any call arriving at FXO 1 > is forwarded to Sip phone 1 only, and any call > received on FXO 2 is transferred to Sip Phone 2, > conversely any call originating from Sip Phone 1 goes > out of FXO 1, and any call originating from Sip Phone > 2 goes out of FXO 2.
You would be doing yourself a great favor by making your first configuration VERY simple: 1 FXO and 1 phone. Walk before you start to run. This method does delay immediate gratification, yet gets one to become a champion sprinter much faster in the long run. Also, your answer is definitely in the archives. Use http://www.google.com/custom?sitesearch=lists.digium.com and http://www.voip-info.org/tiki-index.php?page=Asterisk These are very good resources that keep getting even better. All newbies need to invest some time at these resources. Anon _______________________________________________ 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
