I don't think anyone should make sense of this until you can explain
the call flow, or limit the call trace to a particular call.

Can you do one or the other?
---from the wiki---
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 [email protected]
36 INVITE 2009-12-11T22:17:19 [email protected]

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.

Then use 'sipx-trace' to create an xml file that contains trace data
for messages on your system:

sipx-trace --all-components --output <filename> <token>...
--end wiki snippet---

Thanks


On Sun, Oct 2, 2011 at 2:30 PM, Tony Graziano
<[email protected]> wrote:
> I cantlook at thetrace until later. the itsp must support reinvite in order
> to handle transfers.
>
> the are directions on the wiki to produce a trace for one particular call.
>
> On Oct 2, 2011 2:19 PM, "Ewan McLean" <[email protected]> wrote:
>> Hi
>>
>> I have an autoattendant set up. When someone presses 1 to be
>> transferred, the call rings on the destination phone (and keeps ringing)
>> but the caller gets disconnected. I have isolated this to a particular
>> gateway as the same attendant works perfectly on another gateway
>> provided by a different ITSP. This is the trace file. I'm having
>> difficulty making sense of it as it's a much bigger trace than the ones
>> I've been getting used to recently.
>>
>> On an aside this is the same ITSP I have been having trouble getting
>> outbound calling working with, I haven't had time to look at that
>> properly yet.
>>
>> Can anyone help?
>>
>> Many thanks
>>
>> Ewan
>>
>> --
>> <http://servistech.co.uk>Ewan McLean
>>
>> 02034684428
>> servistech.co.uk <http://servistech.co.uk>
>



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