[asterisk-users] asterisk and sip nat problems

[mike]

thank you very much !
i'll try it asap

.mike


On Thu, 2006-07-06 at 15:37 -0400, Alexander Ginzburg wrote:
> 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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