[Asterisk-Users] asteisk, sip & NAT

[Dan]

Hi,

Have you opened the port 5060 on your firewall? Then you need to open ports
used for RTP, in order to have audio too.
What do you exactly want to do? To call a FWD user when you are connected to
your Asterisk box? To be called by an FWD user?

BR,
Dan


----- Original Message ----- 
From: "Hervé Thibaud" <ht_asterisk at beltegeuse.org>
To: "asterisk-users" <asterisk-users at lists.digium.com>
Sent: Sunday, June 22, 2003 10:13 AM
Subject: [Asterisk-Users] asteisk, sip & NAT


> hi
> My stations are behinds a firewall, the system is windows 2000 & 98, i
> use sjphone
> asterisk is on the internet gateway where is the firewall Shorewall the
> system is linux debian (sid) kernel 2.4.20
> j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy)
> to write my sip.conf but i can't call an external sip user. (an external
> user can call me)
> i try without asterisk with the option proxy 192.246.69.223 port 5060
> but i think rapidely that i have to use proxy adress 192.246.69.247 port
> 5082 and i succeed to call me (and have rings)
> i try to do the same thing i sip.conf but i don't succeed
> where i have to write 192.246.69.247 port 5082 ?
> thanks
>
> ---------------------------------------
> pensée du jour :
> ... c'est pas tout, mais va falloir s'y mettre ...
>
> maître h thibaud
>
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