I am trying to use Asterisk as a "pure" voicemail system and have the following setup: I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the "To:" field, but the INVITE is still being sent to the number asterisk is configured for.
Is there any way that I can configure asterisk to "read" the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number? Thanks Deepak _______________________________________________ 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
