[Asterisk-Users] One way audio - it doesn't make sense

[Jean-Michel Hiver]

> What ports am I missing?  Could the problem be entirely something 
> else?  Somehow I had the feelings that calls going out (since they 
> originate from the device behind the NAT) would not be a problem, but 
> calls coming in could be.
>  
> I really would appreciate a hint from somebody who knows better than I 
> do (i.e. anybody)

Pehaps you have set your device to use an outgoing codec which is not 
supported out of the box by asterisk, such as g.729? ulaw or gsm should 
work. Check your codec config in your sip.conf as well. For debugging 
purposes, you should use ulaw everywhere (assuming your ISP supports it).

Also, are you having any messages on the asterisk command line? Log onto 
your server, type in:

asterisk -r
set verbose 10000000000
set debug 10000000000

And let us know what you're seeing on the CLI.

Cheers,
Jean-Michel.

-- 
Jean-Michel Hiver - http://ykoz.net/
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