On Sat, 2009-12-12 at 11:10 -0500, Geoff Van Brunt wrote:
> >It may be that your gateway is incorrectly detecting a DTMF 0 on the
> >PSTN side and sending it in the RTP.
>
> My thoughts exactly. I believe there is some incorrect DTMF detection
> going on. We have also on occasion had entire conferences booted off,
> but I'm not sure if that is related. We do have 0 assigned to
> receptionist, but their phone does not ring. I wish I could capture a
> trace but it is so intermittent I haven't been able to yet. I can't
> leave debugging turned on either as it causes the voicemail service to
> hang and core dumps. I reported this to the list a while back when
> another user reported them.
Setting the logging to INFO is sufficient for getting a SIP trace, and
should not impact your system significantly. I'd just set it globally
on all the services.
> We are using an Audiocodes Mediant 1000 with a PRI interface. I may be
> able to do a capture from it to Ethereal and hopefully have it
> correspond to a cut off message and see if DTMF is involved. I'm also
> going to have an assistant keep recording messages until they fail and
> maybe get a capture. Is there anything on the wiki about filtering using
> SipViewer?
Have your test caller keep track of the time each call is made (or the
time the voicemail is transferred and work back from that).
Then on your system, run:
sipx-dialog-count /var/log/sipxpbx/sipXproxy.log
it will print a list like this:
Messages Method Time Call-Id
-------------- ---------- -------------------
------------------------------------------------
50 INVITE 2009-12-11T22:22:20
nekyxpqiedpe...@scott.home.skrb.org
36 INVITE 2009-12-11T22:17:19
zbwmxexjtxvl...@scott.home.skrb.org
search that list for the call that starts at the right time (log times
are in UTC), and then use the call-id from that call as the <token>
argument to sipx-trace - you'll get just the one call you're after.
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