[Asterisk-bsd] One Way Audio

[Jay Adelson]

BSD Folks,

Maybe you can help me debug this.  I've got three extensions, configured
essentially identically on Asterisk.  Two are for my Polycom hardphone
(Soundpoint IP 500).  One is a softphone loaded on my PC.

The Asterisk box (NetBSD) is also the NAT and router.

Asterisk binds to 0.0.0.0 without problem (Thanks Tom).

All three extensions register with Asterisk perfectly.

Asterisk registers with Sipphone.com without problems.

A call placed from the softphone out, say to another asterisk installation,
works fine.

A call placed from the polycom out, however, results in a call which the
remote site can hear my side pefectly, but I can't hear them.

Watching logs and traces, packets appear to be flowing fine between the
local Asterisk box and my polycom.

Using a call recorder in my /var/spool/asterisk/monitor directory, recording
the call, both sides are recorded fine, so the problem does appear to be
between Asterisk and the hardphone.

The call remains up until we kill it.

If the remote side calls my phone, particularly with reinvite on, I can
hear the remote side fine and the call works normally.

Has anyone seen this behavior before?  Anything similar?  I've done 
SIP DEBUG and as far as I can understand, it looks like Asterisk thinks
everything is hunky dory.

Thanks,

-j