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

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lucabert at lucabert.de
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Posted: Mon Jun 08, 2015 1:48 am Post subject: [asterisk-users] Almost solved: using my Asterisk from Inter

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 callgroup= pickupgroup= dial=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@lucabert.de) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

asterisk_list at earth...
Guest

Posted: Mon Jun 08, 2015 3:45 am Post subject: [asterisk-users] Almost solved: using my Asterisk from Inter

On Monday 08 Jun 2015, Luca Bertoncello wrote:

Quote:

Hi again, list!

I know, I'm really annoying the list... Smile

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".

Quote:

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 .

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello

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lucabert at lucabert.de
Guest

Posted: Mon Jun 08, 2015 3:53 am Post subject: [asterisk-users] Almost solved: using my Asterisk from Inter

Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:

Quote:

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

Me too...

Quote:

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... Smile

Quote:

-- 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... Sad

Regards
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
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