[Asterisk-bsd] Re: One Way Audio

[Tom Ivar Helbekkmo]

Jay Adelson <jay at adelson.org> writes:

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

Well, since the call runs through an Asterisk that has one foot on the
outside, with real addresses, and one on the inside, with the phones
on, and NAT in between, it's important that canreinvite=no.  Otherwise, 
the phone end points will try to speak RTP directly, and while your
phone is then able to address packets to the real IP address on the
other end, that end can't reach your phone through NAT.

You may also have a codec selection problem.  You should probably
clean out the passwords from a copy of your sip.conf, and post it
here.  At the same time, check which codecs your phones support, and
whether they can be set up with preferences for codec choice.

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145