I never once indicated you should put sipx
directly to the modem. I indicated very precisely what kind of
functionality your should have ina firewall, sipx is not a firewall.
modem
in bridged mode <-> firewall capable of doing nat and symmetrical
(full cone nat), which you can competently configure <--> sipx
I even sent you a sample picture.
Hi folks
Thanks
for the responses. I gave a go at taking my router out of the equation.
Setting it into bridged mode basically turned it into a modem. I then
configured the linux server as a gateway/router, masquerading NAT out
and static NAT in on port 5060. It now seems to be doing outbound and
inbound calls fine, and calls between extensions work. What doesn't work
now that did before, is transfers (full stop), IVRs, voicemail, etc.
even from internal. I'm guessing I've got to add some more iptables
rules beyond just port 5060 but I'm not sure what. And is this the right
way to implement NAT?
Ewan
I've seen very similar results when the ITSP cannot accept SDP without
options as well. Check with your ITSP.
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of Tony Graziano
Sent: Sunday, October 02, 2011 4:29 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Calls dropped during transfer
crap... you need to confirm the nat and port forwarding in your home router
is going to do what it needs to do. unless you can out your broadband into a
bridged mode and control your own firewall properly configured, I would not
waste your time.
On Sun, Oct 2, 2011 at 7:04 PM, Ewan McLean <[email protected]>
wrote:
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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crap... you need to confirm the nat and
port forwarding in your home
router is going to do what it needs to
do. unless you can out your
broadband into a bridged mode and control
your own firewall properly
configured, I would not waste your time.
On
Sun, Oct 2, 2011 at 7:04 PM, Ewan McLean
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:
