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 (lucab...@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