[asterisk-users] asterisk and sip nat problems

[mike]

thank you very much !
i've tryed the rtp range ports definition with iptables with effort,
i'll try this asap !
thank you very much for your time !
.mike

On Fri, 2006-07-07 at 09:49 +1000, Nikolai Lusan wrote:
> On Thu, 2006-07-06 at 16:55 -0400, mike wrote:
> > 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
> 
> My solutions to this was to use siproxd
> (http://siproxd.sourceforge.net/) I run it on the gateway and it
> masquerades the astersik box perfectly. I am running a SIP trunk and
> making and receiving calls over it. Check that out, it probably makes
> things a little cleaner.
> 
>