asterisk users - Feb 2006 - Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *

I don't know what's changed, but four SPA841s and a SPA3000 are no
longer answering when they get an inbound call from *.

This has been a working configuration for weeks.  I *have* been fiddling
with the server config; however, the configuration is under version
control and I've reverted everything to exactly how it was when the
server was working.  Doesn't fix it.  I reset one of the SPA841s to
factory defaults and reconfigured, still has the problem.

Outbound calls from the SPA841s through the * server work fine.

How do I figure out what the SPAs are unhappy enough about to return
404?  Below is a representative SIP DEBUG trace for a call; the OPTION
packet sent due to qualify=yes has the same response.

-Johnathan


Reliably Transmitting (no NAT) to 192.168.1.30:5060:
INVITE sip:192.168.1.30 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b487795;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as24a55bd8
To: <sip:192.168.1.30>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 571b9daf5b03610a33a4c45e05f2235e@192.168.1.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 11 Feb 2006 00:25:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 210

v=0
o=root 703 703 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 19942 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called jcorgan-desk
diamond*CLI>
<-- SIP read from 192.168.1.30:5060:
SIP/2.0 404 Not Found
To: <sip:192.168.1.30>;tag=cb1ee3725d25570ei0
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as24a55bd8
Call-ID: 571b9daf5b03610a33a4c45e05f2235e@192.168.1.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b487795
Server: Sipura/SPA841-3.1.1(a)
Content-Length: 0

Johnathan Corgan wrote:
>I don't know what's changed, but four SPA841s and a SPA3000 are no
>longer answering when they get an inbound call from *.
>
>This has been a working configuration for weeks.  I *have* been fiddling
>with the server config; however, the configuration is under version
>control and I've reverted everything to exactly how it was when the
>server was working.  Doesn't fix it.  I reset one of the SPA841s to
>factory defaults and reconfigured, still has the problem.
>
>Outbound calls from the SPA841s through the * server work fine.
>
>How do I figure out what the SPAs are unhappy enough about to return
>404?  Below is a representative SIP DEBUG trace for a call; the OPTION
>packet sent due to qualify=yes has the same response.
>
>-Johnathan
>
>
>Reliably Transmitting (no NAT) to 192.168.1.30:5060:
>INVITE sip:192.168.1.30 SIP/2.0
>Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b487795;rport
>From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as24a55bd8
>To: <sip:192.168.1.30>
>  
>
There is no username in the above "To" header.  Check your DIAL
command
because something is wrong here.  Thats why you get a 404.  The SPA 
can't match the username.
>Contact: <sip:asterisk@192.168.1.2>
>Call-ID: 571b9daf5b03610a33a4c45e05f2235e@192.168.1.2
>CSeq: 102 INVITE
>User-Agent: Asterisk PBX
>Max-Forwards: 70
>Date: Sat, 11 Feb 2006 00:25:46 GMT
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>Content-Type: application/sdp
>Content-Length: 210
>
>v=0
>o=root 703 703 IN IP4 192.168.1.2
>s=session
>c=IN IP4 192.168.1.2
>t=0 0
>m=audio 19942 RTP/AVP 0 101
>a=rtpmap:0 PCMU/8000
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-16
>a=silenceSupp:off - - - -
>
>
>
>  
>

-- 
Andres