[Asterisk-Users] canreinvite always =no * no matter what we try :-(

[Luki]

> Actully ethereal
OK...

Try canreinvite=yes in the [general] section; this makes it the
default setting for all peers unless specified otherwise. Do the same
for nat=no in [general] to rule out all NAT'ing related issues. You
don't have tT in your Dial() statement, that's good. You say you
verified that no transcoding is needed (i.e. both ends use the same
codec). Well, then it should work!

Once you get it to work, you can individualize the accounts and no
longer use a global setting. But that's down the line.

> asterisk always creates a 'native bridge' and seems to hold on for dear
> life so far as I have seen :-)
It says "Attempting Native Bridge" but it doesn't tell if you if it
succeeded or not; there was once a notice saying the the bridge could
not be established (failed?) but it caused even more confusion. You
could add some statements to the ast_rtp_bridge() code in rtp.c and
give yourself some feedback -- succeeded, failed because X / Y / Z.

Hope that helps...

--Luki