[asterisk-users] Peer is UNREACHABLE

[Luca Bertoncello]

Darryl Moore <darryl at 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 at 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 at 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
(lucabert at lucabert.de)