Connecting Two Asterisk

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Asterisk Users 4 Comments

Hi again!

I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too… I think, my UMTS-Provider doesn’t want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME
configuration!!) installed on my Server…

Well, I will try to configure the Asterisk on my Server to act as “proxy” so that all phones at home talk with my Asterisk at home (now called “wrt”, and my mobile phone talk with my Asterisk on my server (now called “lucabert”).

I followed this HowTo http://sysmagazine.com/posts/125303/ and I got both Servers talking together.

I can call my mobile phone (logged in at “lucabert”) from a phone logged in on “wrt” and a phone at “wrt” from my mobile phone at “lucabert”. Wonderful!

Now the problem: on my phones at “wrt” I can hear what the mobile phone at
“lucabert” sends (with a very good audio-quality), but on this mobile phone I cannot hear a single word spoken with the phone at “wrt”, not even the music on hold I configured…

When I call my mobile phone from a phone logged on at “wrt” I see on the Asterisk at “wrt”:

== Using SIP RTP CoS mark 5
— Executing [4@default:1] Verbose(“SIP/00493511111111-0000001e”, “2,Internal call for Mobile – [00493511111111]”) in new stack
== Internal call for Mobile – [00493511111111]
— Executing [4@default:2] Dial(“SIP/00493511111111-0000001e”, “IAX2/lucabert:MYVERYSECRET@lucabert/00491773333333,,R”) in new stack
— Called IAX2/lucabert:MYVERYSECRET@lucabert/00491773333333
— Call accepted by X.Y.Z.K (format gsm)
— Format for call is gsm
— IAX2/lucabert-1298 is ringing
— IAX2/lucabert-1298 answered SIP/00493511111111-0000001e
— Started music on hold, class ‘default’, on IAX2/lucabert-1298
— Stopped music on hold on IAX2/lucabert-1298
— Hungup ‘IAX2/lucabert-1298’
== Spawn extension (default, 4, 2) exited non-zero on ‘SIP/00493511111111-0000001e’

On the Asterisk at “lucabert” I see:

— Accepting AUTHENTICATED call from A.B.C.D:
> requested format = ulaw,
> requested prefs = (ulaw|gsm|g729|alaw),
> actual format = gsm,
> host prefs = (gsm|g729|alaw|ulaw),
> priority = mine
— Executing [00491773333333@default:1] Macro(“IAX2/lucabert-94”, “stdexten,00491773333333,SIP/00491773333333&DAHDI/1”) in new stack
[Jun 7 21:59:09] WARNING[19888]: app_macro.c:302 _macro_exec: No such context ‘macro-stdexten’ for macro ‘stdexten’
— Executing [00491773333333@default:2] Set(“IAX2/lucabert-94”, “CHANNEL(musicclass)

4 thoughts on - Connecting Two Asterisk

  • Some other data…

    I changed both iax.conf and wrote:

    bandwidth=high allow=all

    Now I see in the log:

    > requested format = ulaw,
    > requested prefs = (),
    > actual format = ulaw,
    > host prefs = (),
    > priority = mine

    and I can hear somewhat, but with a VERY poor quality on my mobile phone… On the other phone however, the quality is very good…

    I’m very very puzzled…

    Thanks for any help!
    Luca Bertoncello
    (lucabert@lucabert.de)

  • Steve Edwards schrieb:

    Yes, I do, but the quality is always very poor…

    Thanks Luca Bertoncello
    (lucabert@lucabert.de)