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

[Michaël Gaudette]

> > 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).

I tried, the only allowed codec in my sip.conf file is GSM, as supported by
my provider.  

My CLI doesn`t show anything special with debug turned on full. Just the
typical:

-- SIP/provider-0154 is making progress passing it to SIP/myid
-- SIP/provider-0154 is ringing
-- SIP/provider-0154 answered SIP/myid
-- Attempting native bridge of SIP/myid and SIP/provider-0154

Mike