[asterisk-users] help with debugging phone call

[Jerry Geis]

Jerry Geis wrote:
> 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)
>
>
>
I enabled sip debug, say it was trying to take "10" instread of 1044. I 
dont have a 10 in my dialplan,
I dont have a _XX in my dialplan... If I put a 10 in my dialplan it runs 
that as expected like playback(demo-congrats).

How do I tell what/what its matching on the 10?

Jerry