[asterisk-users] asterisk and sip nat problems

[Alexander Ginzburg]

in the rtp.conf you specify range of ports, this range should be
forwarded on the firewall to the asterisk box.

I have asterisk running on 192.168.0.250 ip and connect to broadvoice
server.  here is my iptables rule:
-A PREROUTING -i eth0 -s 147.135.0.128 -p udp -m udp --dport 10010:10013
-j DNAT --to-destination 192.168.0.250

with rtp range being:
rtpstart=10010
rtpend=10013

-- Alex.


On Thu, 2006-07-06 at 16:55 -0400, mike wrote:
> Hi all !
> 
> i'm having a strange issue with an asterisk box behind a firewall
> i'm trying to answer a sip call made to an asterisk box with a public ip
> from another asterisk box behind a firewall
> 
> on the natted box i've put
> 
> externip=195.110.XXX.XXX
> localnet=10.1.1.0/255.255.255.0
> 
> and on the phone context i've added
> nat=yes
> 
> 
> the call starts from the natted box, 
> the other phone rings
> on phone pickup, from the public asterisk the message is the following:
> Attempting native bridge of SIP/83.211.XX.XXX-0819b588 and SIP/1-f3d0
> after 3 seconds, the natted box prints the following:
> No one is available to answer at this time
> 
> what do you think it's missing ?
> 
> thank you very much
> .mike
> 
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