Hi! I have three questions:
- I have called from an H.323 softphone to a SIP one, and then I have tryied to transfer the call to be accepted by asterisk. And it has not work. Is it possible to do that? And if it is possible, what I have to do for that works? - In extensions.conf, there's the variable CONSOLE who has Console/dsp value and it says that this is for demos. What are the other two types of Console (Zap/1 and Phone/phone0)? I have to change it to make asterisk operational? - How are prefixes used? What are they made for? Thanks for all your help. Best regards, Mireia _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
