asterisk users - Jun 2015 - Call accepted from not registered peers?

Hi list!

So, new day, new problem...

I tried right now to call from my cellphone a peer in my Asterisk.
The cellphone has correct credentials, but it's NOT registered on my
Asterisk, now.

I just tried to call a peer in my network, from a peer not yet registered.
And it works... :(

The very curious thing is, that I can't find how the call will be
accepted...
Every section in my dialplan has a log, and no log will be displayed on the
CLI...

I just see:

  == Using SIP RTP CoS mark 5
    -- Executing [00493511111111 at default:1]
Dial("SIP/00491773333333-0000000b",
"SIP/00493511111111&DAHDI/1") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/00493511111111
[Jun 11 07:26:04] WARNING[4347]: channel.c:5754 ast_request: No channel type
registered for 'DAHDI'
[Jun 11 07:26:04] WARNING[4347]: app_dial.c:2345 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 66 - Channel not implemented)
    -- SIP/00493511111111-0000000c is ringing
  == Spawn extension (default, 00493511111111, 1) exited non-zero on
'SIP/00491773333333-0000000b'

I tried to remove ALL includes in my [default], leaving just a log, but
it calls, too...

My [default]

exten => _X.,1,Verbose(2,DEFAULT)
include => internal_calls
include => luca_incoming
include => fax_incoming
include => anika_incoming
include => messagenet_incoming
include => myproxy

What's wrong, now?
Many thanks for your help!

Luca Bertoncello
(lucabert at lucabert.de)

On 2015-06-11 00:31, Luca Bertoncello wrote:
> Hi list!
> 
> So, new day, new problem...
> 
> I tried right now to call from my cellphone a peer in my Asterisk.
> The cellphone has correct credentials, but it's NOT registered on my
> Asterisk, now.
> 
> I just tried to call a peer in my network, from a peer not yet 
> registered.
> And it works... :(
> 
> The very curious thing is, that I can't find how the call will be 
> accepted...
> Every section in my dialplan has a log, and no log will be displayed on 
> the
> CLI...
> 
> I just see:
> 
>   == Using SIP RTP CoS mark 5
>     -- Executing [00493511111111 at default:1]
> Dial("SIP/00491773333333-0000000b",
"SIP/00493511111111&DAHDI/1") in
> new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/00493511111111
> [Jun 11 07:26:04] WARNING[4347]: channel.c:5754 ast_request: No
> channel type registered for 'DAHDI'
> [Jun 11 07:26:04] WARNING[4347]: app_dial.c:2345 dial_exec_full:
> Unable to create channel of type 'DAHDI' (cause 66 - Channel not
> implemented)
>     -- SIP/00493511111111-0000000c is ringing
>   == Spawn extension (default, 00493511111111, 1) exited non-zero on
> 'SIP/00491773333333-0000000b'
> 
> I tried to remove ALL includes in my [default], leaving just a log, but
> it calls, too...
> 
> My [default]
> 
> exten => _X.,1,Verbose(2,DEFAULT)
> include => internal_calls
> include => luca_incoming
> include => fax_incoming
> include => anika_incoming
> include => messagenet_incoming
> include => myproxy
> 
> What's wrong, now?
> Many thanks for your help!
> 
      It does not matter that your phone is not registered with Asterisk. 
  As long as it has the proper credentials it will be able to send calls. 
  You only need to register if you want to RECEIVE calls on that phone.

-- 
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