VoIP Mailing List Archives

[lucabert at lucabert.de]

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)

--
_____________________________________________________________________
-- 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

[cursor at telecomabmex...]

On 2015-06-11 00:31, Luca Bertoncello wrote: