Re: Multiple phones through a 2Wire 2700 HGV (Bt Business Hub)
On Jan 10, 6:28 pm, alexd <troffa...@hotmail.com> wrote:
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>
> Perhaps it's worth replacing the BT router, as you may run into a similar
> problem again in the future with other applications.
>[/color]
I have considered that, but have heard a of number of instances where
a third party router would not work correctly with a BT ADSL
connection.
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>
> If I read you right, the plan is:
>
> (Net)--(BT router)--(Another router)--(switch)--(handsets)
>[/color]
Yes That's correct.
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>
> I think before you throw any more hardware at the problem, you should
> validate that what you're planning is going to work. I can't see how
> adding another link in the chain is going to fix a firewalling problem on
> the BT router. If you just use the one handset, and modify the rules to
> allow it out, does it work? How does internet browsing work if you have
> to add an explicit permit rule to allow a host out, but can only add rule
> at a time?
>[/color]
I may be wrong but it appears that the only way to turn the firewall
off on this router
is to assign whatever you are connecting to its internal interface to
the DMZ, to which it then assigns
an external (dynamic) IP, and as it only allows you to do this to one
host, this will have to be a router
with an inside and outside ethernet interface so that I can assign non
routable internal IP's to the phones.
It is incoming traffic which the BT Firewall is blocking - not 5060
but the RTP range 10000 - 12000
We can initiate and answer calls and register the handsets but lose
audio.
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>
> Have you tried using a handset [or softphone] from home to test it?
>
> Does Asterisk have a public IP? If not, have you told it what it's public
> address is? [[url]http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP[/url]
> +externip]
>
> Are the handsets SIP or IAX?[/color]
SIP - a mixture of ATCOM AT530 and Seimens S450IP
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> Have you tried disabling/enabling SIP fixup on the PIX?[/color]
No need, the Pix end of things is quite happy.
The Asterisk Server has a public IP, and connections to it from other
sites we run
have no problems at all - we have a satellite office with its own
Asterisk and the
two are connected by IAX. We have a further site in france with
multiple phones on an
ADSL from wanadoo.fr which again connects to the main asterisk site
with no problems.
At home I have a sip phone which sits behind a BT Router with Static
IPs and it works fine.
It is just this site - and this router - which are the problem.
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> If you have to add another router, you'd probably be best off adding
> something that can terminate a VPN from the PIX, and run the calls over
> the VPN. This would bring all the usual benefits of VPNs with it.[/color]
I have a spare PIX 501 which I was thinking of using as the router,
which would mean I could
possibly use VPN, but on voip to voip calls wouldn't that effectively
stop RTP from bypassing the asterisk?
As I understand it, Asterisk initiates the connection but then hands
it off to the two hosts using RTP for the
voice and SIP for the call control. If I am wrong, I'm sure you'll let
me know :-)
btw I do appreciate the time you are taking to try and help - I'm
sorry if I haven't explained things clearly.
Alister
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