That's sad. I'll start talking to my ISP. Thanks Bob
On Thu, 2009-04-02 at 16:07 -0400, Matt Sales wrote: > Bob, you're dealing with a double NATted environment with both your > Asterisk server and phones behind firewalls. Personally I think this > is going to do nothing but give you headaches. I would add a 2nd NIC > card to your asterisk server and assign it a public IP address. Now > just have your softphones and hardphones register to the public IP > address. You will still need to set nat=yes for those remote > extenstions in users.conf > > I'm sure others will chime it that this a major security risk but if > you use iptables to block all ports except those required by asterisk > (SIP, IAX, & RTP) and set secure sip passwords you should be ok. > > > On Thu, Apr 2, 2009 at 2:57 PM, Bob Crandell <[email protected]> > wrote: > > I have NAT=yes through out. > I have both externip and localnet set. > > > > > On Thu, 2009-04-02 at 00:03 -0700, Marvin Whitfield wrote: > > > Bob, > > Definately a NAT issue. Had the same issues when I first started > > setting up remote extensions. Look in your sip.conf for externip=... > > and localnet=.... You'll use the to tell asterisk what's internat > and > > what's external. Also remember to at NAT=yes to extensions that will > > be remote. The last thing you might need to modify is your firewall > > and phone settings. These vary widely by manufacturer but the > general > > idea is that you need asterisk to be able to reach the phone that > will > > be behind a NAT firewall so you'll probably need to forward some > ports > > or configure the phone to keep the connection alive. Hope that > helps. > > > > -- > > Marvin > > > > On 4/1/09, Matt Brown (HC) <[email protected]> wrote: > > > Hi Bob, > > > > > >> Hi guys, > > >> > > >> I've been testing softphones in Windows and SuSE. From outside > the > > >> office, I can make a call to a different extension and to an > outside > > >> number but neither the caller nor the callee can hear anything. > I > > >> thought I was being stupid until I bought a Grandstream GXP 2020 > > >> which wouldn't speak to me either from outside the office. It > does > > >> work inside the office. I tested the softphone inside the office > > >> and it works too. Ok so it's the firewall. I have port 5060 UDP > > >> and 10001-20000 UDP open and pointed to the Asterisk box. What > am I > > >> missing? I'm getting so close to being able to go live with this > > >> thing. > > >> > > > > > > Just double check the /etc/asterisk/rtp.conf to make sure port > range > > > being used is correct. In addition I agree with Matt that it > sounds > > > like a NAT issue. Take a peek at sip.conf or users.conf for the > > > extension in question and check nat=yes is set for that > user/extension. > > > > > > Regards > > > > > > Matt Brown > > > > > > _______________________________________________ > > > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > > > > > asterisk-gui mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-gui > > > > > > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-gui > >
_______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui
