Hello, Asterisk does not act as a SIP Proxy as you may have in mind. Each call is treated independently, that is - codec capabilities of one call don't go to the other one during a reinvite. Only the IP address and Port go.
Joshua Colp -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, September 19, 2005 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] T.38 & Canreinvite (yes, again) I know this has been asked before, but I've checked the archives and I haven't found anybody that has given a definitive yes or no, just "yeah, it should work.....". If I have a T.38 gateway like a Cisco 5300 and a T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work? I have it setup and it doesn't work, so I want to know if I am doing something wrong, or if it just won't work. If I make a voice call, I see the media stream go from the gateway to the ata directly. When I fax, I see the stream go that way as well, but it is g.729. I see INVITE messages from my ATA that reference T.38, but they go to the * box, not the gateway and therefore * ignores it. Any thoughts? PA _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
