[Asterisk-Users] Avaya 4620 hardphone

[Tom Lynn]

I've just interfaced an Avaya 4621 set to *@home.  It's running the
2.2 SIP firmware released August 17th.  

I've run into some strange behavior with this phone.  1st, in order to
get two way audio, I had to tell * that it was behind a NAT even
though it is on the same subnet as the * server.

Second, it doesn't seem to be able to dial any extensions or features
that use the * digit in them.  The phone remains off hook, and I get a
fast busy after about thirty seconds.  I also can't find any log
entries that shoe the attempt was even seen by the server.  I've
completely disabled the internal dial plan inside the phone and the
problem persists.

Third, the mute button seems to disable audio in both directions, not
simply the local microphone.

Has anybody else interfaced one of these phones to *?  If so, did you
experience these same problems and did you find any solutions?

I believe this is the first public release of SIP firmware for this
phone and I know it hasn't been out that long.

Thanks - Tom