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@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@lucabert.de)