[asterisk-users] Connecting two Asterisk

[Luca Bertoncello]

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 at default:1] Verbose("SIP/00493511111111-0000001e", "2,Internal call for Mobile - [00493511111111]") in new stack
  == Internal call for Mobile - [00493511111111]
    -- Executing [4 at default:2] Dial("SIP/00493511111111-0000001e", "IAX2/lucabert:MYVERYSECRET at lucabert/00491773333333,,R") in new stack
    -- Called IAX2/lucabert:MYVERYSECRET at 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 at 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 at default:2] Set("IAX2/lucabert-94", "CHANNEL(musicclass)=default") in new stack
    -- Executing [00491773333333 at default:3] Dial("IAX2/lucabert-94", "SIP/00491773333333,,R") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 00491773333333
    -- SIP/00491773333333-00000008 is ringing
    -- SIP/00491773333333-00000008 is ringing
    -- SIP/00491773333333-00000008 is ringing
    -- SIP/00491773333333-00000008 answered IAX2/lucabert-94
  == Spawn extension (default, 00491773333333, 3) exited non-zero on 'IAX2/lucabert-94'
    -- Hungup 'IAX2/lucabert-94'

Well, I'm very puzzled...
Can someone help me?

Thank you very much!
Luca Bertoncello
(lucabert at lucabert.de)