It's a SOHO installation, when I say home router I just mean it's not a
Cisco enterprise router or anything, it's a broadband router. Sipx is in
the home office. Each computer is setup with security and IPTABLES will
be added to sipx once it's working.
Tony Graziano wrote:
hold on a sec... what is the relevance of the home router type mentioning?
is sipx at home or somewhere else? if so, what type of firewall is being used?
On Sun, Oct 2, 2011 at 6:47 PM, Ewan McLean
<[email protected]> wrote:
It should be just one call - I rotated out the logs before I placed a test
call then merged them immediately afterwards? I dialled in on the ITSP
trunk, got connected to my auto attendant, pressed one to transfer to an
internal extension (201), heard the 'hold while i transfer you' blurb, then
got cut off. Meanwhile extension 201 started ringing even though the call
had been cut off, it was silence when the extension was picked up.
Intranet domains *.servistech.co.uk
Intranet subnets 192.168.1.0/25
Sipx at 192.168.1.100
Extension 201 on a polycom at 192.168.1.5
Home type ADSL router/switch at 192.168.1.100 with no port forwarding,
firewall etc. set up as far as I can tell.
The ITSP is called Numbergroup at sip.numbergroup-services.com
I tried an unmanaged gateway but it didn't seem to work. I plugged in the
same address as the destination but there was nowhere to plug in
authentication details and as far as I know it's setup to require those.
I'm new to the raw SIP messages here, I used to use asterisk which hid all
of it. Happy to learn, but it's a bit overwhelming. I'm not sure what the
difference between the media server, bridge, proxy, registrar, trunking etc.
components are.
Does any of this help?
Tony Graziano wrote:
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?
The ITSP is syaing the options are not acceptable, stating: Reason:
Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Which probably means they dont like the codec in the media server.
That really shouldn't matter since the destination should NOT be the
media server. The REFER is not being handled locally from what I can
see, otherwise they would be accepting the call.
Explain what you have in intranets/subnets and what type of network
sipxecs is on locally. Then provide how you connect to the ITSP
(registration or IP based).
In any case, I think I see the reinvite in frame 36, and the ITSP says
BYE at frame 47 saying an incompatible destination.
So my question here is, since it appears that the ITSP (see frame 10)
supports REFER, why don't you set them up as an unmanaged gateway and
esentially leave sipxbridge out of the mix?
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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