[asterisk-users] help with debugging phone call

[Jerry Geis]

I am running 1.4.22.

I am doing a simple call into the dialplan and am getting a strange error:

[Nov  3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: 
Failed to authenticate user "404" 
<sip:404 at 192.168.1.8>;tag=547521CB-DB0D6130

This is the only line that prints on the console...

Typically I get a few lines like:
    -- Executing [33 at smvoice-sip:1] Dial("SIP/404-18afe560", 
"SIP/bt610tmm/1044") in new stack
    -- Called bt610tmm/1044
    -- SIP/bt610tmm-b4046c70 answered SIP/404-18afe560
    -- Packet2Packet bridging SIP/404-18afe560 and SIP/bt610tmm-b4046c70
  == Spawn extension (smvoice-sip, 33, 1) exited non-zero on 
'SIP/404-18afe560'


Note that both call attempts are from the same phone 404.
How can I find out why the first situation above is not showing me 
dialplan messages like case number 2 above
and debug this situation?

THanks,

Jerry

------------------ extensions.conf

[smvoice-sip]
; case 2 above
exten => 33,1,Dial(SIP/bt610tmm/1044)

; case 1 above
exten => 1044,1,Dial(SIP/bt610tmm/1044)