19 Dec
2003
19 Dec
'03
4:29 p.m.
Jiri,
The tcpdump is at http://www.columbia.edu/~alan/ser.tcpdump
Here's the corresponding ser syslog:
Dec 19 16:16:38 ren kernel: eth0: Setting promiscuous mode.
Dec 19 16:16:38 ren kernel: device eth0 entered promiscuous mode
Dec 19 16:16:56 ren /usr/sbin/ser[5693]: ACC: call missed:
method=INVITE,
i-uri=sip:17548@128.59.39.127;user=phone;phone-context=private,
o-uri=sip:alan@68.198.11.112:5060,
call_id=798FB767-319F11D8-B06B89D2-F2E2B15E(a)128.59.59.242, from="43754"
<sip:43754@128.59.59.242>, code=408 Request Timeout
Dec 19 16:16:56 ren /usr/sbin/ser[5693]: redirection to voicemail
Dec 19 16:16:56 ren /usr/sbin/ser[5676]: ERROR: t_should_relay: status
rewrite by UAS: stored: 408, received: 487
Dec 19 16:16:56 ren /usr/sbin/ser[5681]: ERROR: t_should_relay: status
rewrite by UAS: stored: 408, received: 487
Dec 19 16:17:19 ren kernel: device eth0 left promiscuous mode
The timestamp in the log corresponds to the 487's coming back from the
7960 phones.
Asterisk is running on 128.59.39.163 in this trace... (Ignore the
couple of transactions from some failed REGISTERs for phones that we
haven't configured into ser yet.)
/a
Alan Crosswell wrote:
Jiri Kuthan wrote:
At 08:58 PM 12/19/2003, Alan Crosswell wrote:
Both UAs are cisco 7960's (POS3-04-3-00). I don't see why either would
time out. It's only 15 seconds for fr_inv_timer. I also just tested
with another singly-registered user and the error still happens so it's
not a multiple dset thing.
I'm trying to do failure route to voicemail
(which is working) but
this error is logged:
ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
I googled this and see that this came up last in October
(http://lists.iptel.org/pipermail/serusers/2003-October/002921.html)
but I don't see any evidence of a solution in the thread.
I think it is a msitake in log severity and it should be softened to a
warning.
What possibly happens is that a UAS times out at the same time when
the call
is canceled, and it sends 408 timeout and milisecond later a 487 in
response
to CANCEL... What UAS did you use?
Generaly, such issues are easier to capture if you keep track of all
SIP passing
your site. I think that's a very good practice. If you get me message
dumps,
I will try to verify what really happened.
Part of the problem is since asterisk is running on localhost I won't be
able to get captures for that. I suppose I could move it to another
machine just to help get some tcpdumps. I'll do that and send along.
Regarding voicemail: SEMS supports voicemail2email. As the number of
people
interested in IVR grows, there is now a first IVR version of voicemail
too
-- but this piece of work has never been really tested.
I'll be happy to test it, but you need one of the canned IVR messages to
be "Weasels have eaten our phone system" before you'll see a mass exodus
from asterisk to sems;-)
-jiri
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