[Asterisk-Users] ICH PROBLEM

[listas iPfone]

Hi !

I´m using * with a snom 200 phone, i can use FWD but cant use ICH.

Someone can tell me if my setup is correct?

sip.conf:

register =>user:pass at sipauth.deltathree.com/33

extensions.conf:

exten => _7X.,1,Dial(SIP/${EXTEN:1}@sipauth.deltathree.com)

In CLI the registration is ok but when i try ex. 7551136752312 nothing
happens, i get a forbiden message.

Thanks

Miklos

----- Original Message ----- 
From: "Olle E. Johansson" <oej at edvina.net>
To: <asterisk-users at lists.digium.com>
Sent: Monday, September 29, 2003 8:26 AM
Subject: Re: [Asterisk-Users] IAX and NAT


> Brancaleoni Matteo wrote:
>
> > VoIP protocols normally use 2 connection:
> > * 1 for control (eg on port 5060 for sip)
> > * 1 for the RTP (media stream)
> > The latter hasn't a fixed port, since is negotiated
> > by the control connection. That could cause some troubles
> > with NAT & firewalls.
> >
> > IAX doesn't use 2 ports, but only one .
> > So on the same port it brings the control connection &
> > the RTP stream. So NATting IAX isn't a problem
> >
> Also, IAX is client-driven, the IAX client opens a channel to the server
and
> keeps it open for calls both ways.
>
> SIP is a peer-to-peer protocol and a phone needs to be able to receive
incoming
> calls. If the phone, or the "SIP UAC/S (User agent client/server)
software",
> is behind a NAT, there's no way any phone out there can reach it on the
inside.
>
> There are a lot of fixes, ranging from using a SIP proxy on the outside
for incoming
> calls and keeping a NAT session open with fixes called "NAT pings" to
protocols
> that opens up port forwarding from the NAT to the inside client (UPNP) and
protocols
> that let the client investigate the NAT situation (STUN) and be more
clever.
>
> The long term fix is to remove NAT boxes and use IPv6 or allocate more
IPv4 addresses
> ...or, as some people on this list advocate, use another protocol.
>
> /Olle
>
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