asterisk users - Jun 2015 - Almost solved: using my Asterisk from Internet

Hi again, list!

I know, I'm really annoying the list... :)

Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails)
to accept my mobile phone from Internet.
It was a problem with the network and the firewall.

Now I can log my mobile phone in my Asterisk in and the phone is  
REACHABLE. Wow! Got it!

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...

Here what I did (following this article:  
http://www.linuxjournal.com/article/9399):

sip.conf:

localnet=192.168.200.0/255.255.255.0
localnet=192.168.20.0/255.255.255.0
externhost=mypc.noip.com
externrefresh=180

rtp.conf:

rtpstart=10000
rtpend=10100

users.conf:

[00491773333333]
fullname = 00491773333333
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=yes
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgrouppickupgroupdial=SIP/00491773333333

And on my Firewall:

/sbin/iptables -t nat -A PREROUTING -i ppp0 -p udp -m udp --dport  
10000:10100 -j DNAT --to-destination 192.168.20.120
/sbin/iptables -t nat -A PREROUTING -i ppp0 -p udp -m udp --dport 5060  
-j DNAT --to-destination 192.168.20.120

Any idea, what can be wrong now?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)

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

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>:
>> 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.
Me too... :)
> 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
Well, PSTN (and ISDN) in Germany will be shutted down in short...  
That's why I'm experimenting with Asterisk now... :)
> -- 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.
I wanted to write the list again, since I maybe got it just adding  
"allow=all" for the user...
I'll try your configuration this evening, too and report to the list...

I'm very happy, that now I can login in my Asterisk at home and I  
don't need another Asterisk on a separate server.
Firewall can be very difficult to setup, sometimes, for a SysAdmin as  
I be, too... :(

Regards
Luca Bertoncello
(lucabert at lucabert.de)