Darryl Moore <dar...@moores.ca> schrieb: > I think your phone may be trying to register with the username '1234', > while your sip configuration is expecting 'luca'. Can you try changing > your phone registration credentials to use 'luca'? Can you give us a sip > transcript when you try to place a call from it?
Well, another information (then I **MUST** go sleep...): I tried to use my mobile phone logging to my "own Asterisk" with the login data of my wife's telefon. Now this user is REACHABLE... So I think, it was a problem on her phone... I can't call and receive calls. I think, that it's a problem of my Dialplan. If I try to call the mobile phone from AsteriskNOW (later: "the world"), I see that in Asterisk's log ("my own Asterisk"): == Using SIP RTP CoS mark 5 [May 29 00:07:49] NOTICE[1106]: chan_sip.c:20163 handle_request_invite: Call from '00493511111111' to extension '00493512222222' rejected because extension not found. That's very strange, since I call from Twinkle and it has the number "1234"... If I call my mobile phone using my VoIP-phone (connected on the same "my own Asterisk") I get that: == Using SIP RTP CoS mark 5 == Call from 00493511111111 to 00493512222222 == Outgoing using pbxluca == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 [May 29 00:09:25] WARNING[1106]: chan_sip.c:12800 check_auth: username mismatch, have <00493511111111>, digest has <00493512222222> [May 29 00:09:25] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "00493511111111" <sip:00493511111111@172.16.34.132>;tag=as058adbf2 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 00493512222222, 9) exited non-zero on 'SIP/00493511111111-00000004' Maybe this is the same problem, since I didn't configured my own Asterisk to manage "internal calls" (since I don't need to call my wife on VoIP... :D) And, last but not least, if I try to call from my mobile phone Twinkle I get this: == Using SIP RTP CoS mark 5 == Call from 00493512222222 to 1234 == Outgoing using pbxanika == Using SIP RTP CoS mark 5 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 1234, 15) exited non-zero on 'SIP/00493512222222-00000006' And if I try to call my VoIP-phone I get that: == Using SIP RTP CoS mark 5 == Call from 00493512222222 to 00493511111111 == Outgoing using pbxanika == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 [May 29 00:12:02] WARNING[1106]: chan_sip.c:12800 check_auth: username mismatch, have <00493512222222>, digest has <00493511111111> [May 29 00:12:02] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "00493512222222" <sip:00493512222222@172.16.34.132>;tag=as193c26b0 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 00493511111111, 15) exited non-zero on 'SIP/00493512222222-0000000a' Maybe can these information help someone helping me? Thanks a lot! 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