[asterisk-users] help with debugging phone call
geisj at pagestation.com
Mon Nov 3 09:46:09 CST 2008
Jerry Geis wrote:
> I am running 1.4.22.
> I am doing a simple call into the dialplan and am getting a strange
> [Nov 3 08:32:27] NOTICE: 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
> 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?
> ------------------ extensions.conf
> ; 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?
More information about the asterisk-users