Hi Jonas,

try to put the follow in /etc/asterisk/sip.conf

bindport=5060
bindaddr=0.0.0.0
externip="your_public_ip"
localnet=192.168.1.0/255.255.255.0
nat=yes

replace "your_public_ip" and 192.168.1.0 with the right values (in localnet you have to insert lan address (eg. 192.168.1.0) and not server address (eg. 192.168.1.211)

then nat on your router/firewall (server side) towards your asterisk server
5060(UDP)
10000-2000(UDP)

Then configure your sip client as the follow:
server "your_public_ip"
port 5060

obviously with your extension number and related password.

Let me know.

Regards,
Marco




Il 15/01/2010 16:23, jonas kellens ha scritto:
Hello Marco,

to keep the NAT tunnel open between my SIP-phones behind NAT/firewall I send SIP-option packets from my SIP-server (Asterisk) to the SIP-phones.

But when I restart my SIP-server and my firewall, my SIP-server complaints :

[Jan 14 16:50:26] WARNING[1480]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1e56c160 (len 552) to publicip:5063 returned -1: Operation not permitted
[Jan 14 16:50:26] WARNING[1480]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1e56c3d0 (len 546) to publicip:5062 returned -1: Operation not permitted
[Jan 14 16:50:26] WARNING[1480]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1e562440 (len 546) to publicip:5061 returned -1: Operation not permitted

So the SIP option packets do not get through my Endian firewall any more.

That's why I need to just open up this range 5060 --> 5063.

Jonas.

On Thu, 2010-01-14 at 21:35 +0100, Marco Biovi wrote:
Hi,

is not necessary to open ports on firewall or nat something.
Connection will be from sip client (phone) to sip server, I have the same situation and all my ip phone use the 5060 port.
In every phone you will set server ip, server port, domain, realm, ... All this parameter refers to server.

I hope this can help you.

Regards,
Marco
    

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