I don't think you can do that. You can easily issue a 302 with something like SER or OpenSER. I believe the only thing Asterisk can do is receive a call on the initial URI and open a channel to the destination and connect them. Media could pass directly between those two points but your Asterisk box would still have to participate in the signaling. Think of Asterisk as a B2BUA instead of a SIP call router/response system.
-------------------------------------------------- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Thursday, May 03, 2007 6:18 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE >From: "CSB" <[EMAIL PROTECTED]> >Date: Thu, 3 May 2007 21:51:02 +1200 > >I want to get Asterisk to redirect an incoming SIP INVITE to another SIP >URI. I was looking at the Transfer application but it seems to You may want to elaborate the requirement. How is the incoming INVITE initiated? Is the originator a user in your system? Does the other URI represent a peer? etc. Yuan Liu >be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). >Is there an alternative way to do this on Asterisk 1.2.18? > >Regards > >Cameron _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users