[asterisk-users] Almost solved: using my Asterisk from Internet

[A J Stiles]

On Monday 08 Jun 2015, Luca Bertoncello wrote:
> Hi again, list!
> 
> I know, I'm really annoying the list... :)

Everyone has to start somewhere; and at least you aren't asking hundreds of 
questions in one go, including some which come under the heading of "Don't 
even think about trying to set this up until you have got X working", then 
ignoring every answer you received and doing something totally different.  
That's "annoying the list".

> If I call a phone at home using my cellphone it works and the quality
> is perfect!
> If a phone at home call my cellphone, however, the quality on my
> cellphone is very poor, but on the other phone is perfect...
> 
> I think, it is something by the codecs, but I don't know what...

Codecs would be the first thing I would be looking at.

The "native" codec used by the PSTN throughout Europe is G.711 A-law, or just 
alaw for short; and if you are making a system which connects with the PSTN, 
there is rarely a good reason to use anything else; since something, somewhere 
-- and most probably *your* Asterisk server -- is going to wind up having to 
translate from one codec to another.  That is going to  (1)  take a finite 
amount of time and  (2)  introduce distortion.

Try, in the top section of your sip.conf file,

disallow=all
allow=alaw

And that ought to fix it.

If in any doubt, add NoOp() statements at strategic points within your 
dialplan so as to show the value of the channel variable ${SIP_CODEC} .  


-- 
AJS

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