[asterisk-users] SIP/NAT 1.2 1.4 questions

[randulo]

I now have both 1.2 and 1.4.4 boxes.

Each asterisk is behind NAT on a fixed ip with all the externip,
nat=yes, and forwarded ports etc set up.

I have two multiline SIP phones, Linksys 941 and a Polycom ip500.
THese both work normally on the 1.2 box. I took the exact configs from
sip.conf and copied them to the 1.4 box. The phones remain UNREACHABLE
to the 1.4 box, but they can call out.

The other odd behavior is that the Linksys can call the echo test
extension, I see the answer and the echo application but there is no
sound. If I playback a file before the echo test, audio passes.

Can anyone see a reason why the 1.4.4 box would work differently on
the same phones? And why a playback() helps start audio for the echo
test?