[asterisk-users] Authentication failure

[Adrian Marsh]

Hi I'm having trouble routing a call between two A*k servers I admin.


SERVER- A: has a simple extensions set, and just needs to Dial to server
B, but authenticate as part of the dial:

 

exten =>
777,1,Dial(SIP/abc-777:mypassword at someip.no-ip.info:5071/777,40,trw)

exten => 777,2,Hangup

 

So that should pass the call to the server listening on port 5071 of
someip.no-ip.info, using the username of abc-777 and password of
"mypassword", and pass it into extension 777 on that server.

 

SERVER-B has a sip account defined as:

 

[abc-777]

type=friend

secret = mypassword

context = local

host = someotherip.no-ip.info

;disallow = all

;allow = ulaw

canreinvite = no

nat = yes

qualify = no

 

If I run a 'show peer abc-777' then I get a peer ID through ok.

 

But if I try to place a call, I get the following message on SERVER-A
and the call disconnects.

 

 

    -- Executing [777 at from-sip-UK:1] Dial("SIP/ADRIANSPHONE-09dd5178",
"SIP/abc-777:mypassword at someip.no-ip.info:5070/777|40|trw") in new stack

[2011-04-12 15:18:03] WARNING[13926]: channel.c:720 ast_best_codec:
Don't know any of 0x4000 formats

    -- Called abc-777:mypassword at someip.no-ip.info:5070/777

[2011-04-12 15:18:03] NOTICE[17058]: chan_sip.c:12108
handle_response_invite: Failed to authenticate on INVITE to '"Adrian
Marsh" <sip:ADRIANSPHONE at 82.XXX.XXX.26>;tag=as0ff33d62'

    -- SIP/someip.no-ip.info:5070/777-09b16048 is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

    -- Executing [777 at from-sip-UK:2] Hangup("SIP/ADRIANSPHONE-09dd5178",
"") in new stack

  == Spawn extension (from-sip-UK, 777, 2) exited non-zero on
'SIP/ADRIANSPHONE-09dd5178'

 

 

If I turn traces on, on SERVER-B I see the line:

 

Found peer 'abc-777'  so I think the peer is authenticating ok.

The context of the user looks right for accessing extension 777 on
SERVER-B.

 

 

Thanks,

 

Adrian

 

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