Hi list, I have an asterisk server with a Zap 4 port FXO card connected to the PSTN, all of the clients are SIP softphones (I have tried LIPZ4 and kphone). I have successfully configured asterisk to route incoming calls from the PSTN to an extention that is a SIP softphone and the user can answer the call. Question I have is how do I call a PSTN address or extention number from a SIP softphone if the number to dial needs to be a SIP address in the client software? I assume that you need to encode the number or extention into a SIP address, specifying the IP address (or hostname) of the asterisk server as the recipient. Is this correct? And how do I encode the addresses? Is there anything special I need to configure on the asterisk side to act as a media gateway (over and above the basic config I found in the document "How to configure asterisk for LIPZ4")?
Thanks for the help. Dave _______________________________________________ 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
