On Fri, 11 Jan 2008 11:28:51 -0800, Alister wrote:
[color=blue]
> On Jan 10, 6:28 pm, alexd <troffa...@hotmail.com> wrote:[/color]
[color=blue]
> 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,[/color]
[color=blue]
> 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.[/color]

[url]http://www.dslreports.com/forum/2wire[/url]

There are some 2Wire experts in there, might be worth a shot if you're
reluctant to bin it.
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>[color=green]
>> 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[/color]

OK here's another idea - how about putting the IAX firmware on the
Atcoms? Won't fix the Siemens, of course.
[color=blue]
> It is just this site - and this router - which are the problem.[/color]

Replace the router. It can't be that hard, all you need is username,
password and the static IP details [if you've got them]. Having googled
your router, I'm concerned that there is a VoIP implementation on there,
and it may be doing silly stuff to your SIP traffic.
[color=blue][color=green]
>> 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?[/color]

Yes. Calls will be fine from the branch to the site where Asterisk is,
but calls from said branch to other sites over SIP will again be one
sided. If you've got enough bandwidth at the Asterisk end, you could stop
the relevant extensions from being able to reinvite and you should be OK.
[color=blue]
> 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 :-)[/color]

[url]http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite[/url]

explains how Asterisk handles re-invites.


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