[asterisk-users] Interesting One Way Audio

[Thermal Wetland]

I have an Asterisk box, 1.4.30 with a PRI.

A Mitel 3300 is connected to the Asterisk box via SIP trunking.

When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.

But - when I route the Mitel user to echo() it works, send and receive.  The
Mitel user also can record and playback greetings.

One thing I have noticed is that when the Mitel user dials a number that
autoanswers line 1-800-555-1212 the Mitel user will hear audio for 1/2 a
second then it is dropped.

I turned of iptables and it acts the same way.

Anyone have any ideas?

-- 
-Thermal
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