[Asterisk-Users] T.38 & Canreinvite (yes, again)

[Joshua Colp - Asterlink]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
info at beprojects.com
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
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