I'm using this scenario & have had no problems.
When you say that HT-286-2 begets silence, do you mean it does not ring,
or that when you pick up after the ring there is no audio?
-g
On Fri, 2004-06-18 at 14:31, Nathan Martinez wrote:> I have 2 Grandstream HT-286 devices and an Asterisk server. The *
> Server is not using NAT and has port 5060 opened up. One HT-286 is
> using traditional NAT and the other HT-286 is behind a residential DSL
> router/firewall. I have the HT-286 setup as the "DMZ Host" in
the
> router/firewall so that all incoming connections are forwarded to the
> HT-286.
>
> HT-286-1 ====== NAT FW ====== * Server ======= Router/FW ====== HT-286-2
>
> In the setup for HT-286-2 , I have filled in the "Use NAT IP"
field with
> the public IP for that location. I did the same thing for HT-286-1 and
> then I mapped a public IP to its private IP in the NAT FW. At this
> point, the two devices can call each other without any problems.
>
> I want to use the HT-286 for our traveling users who will never know
> what their IP is. When I remove the "Use NAT IP" entry on
HT-286-1 as
> well as remove its direct IP mapping from the NAT FW, HT-286-1 can
> register with the * Server, but when I try to call HT-286-2, all I get
> is silence. If I do a 'sip show channels' it shows that the call
is
> connected. Here is what I have in my sip.conf for these two units:
>
> [305]
> type=friend
> host=dynamic
> nat=yes
> qualify=100
>
> [307]
> type=friend
> host=dynamic
> nat=yes
> qualify=100
>
> Has anyone used these units in this scenario? Does anyone have any
> hints as to what I can try to get this working? Your help is much
> appreciated.
>
> Nathan Martinez
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users