On Sat, Mar 19, 2011 at 2:10 AM, edward choi <[email protected]> wrote: > My Asterisk server is behind a NAT and I have set: > ---------------------------------------------------------------------------- > externhost="my.server.address" > externrefresh=180 > localnet=192.168.0.0/255.255.0.0 > localnet=10.0.0.0/255.0.0.0 > localnet=172.16.0.0/12 > nat=yes > --------------------------------------------------------------------------- > in [general] section of sip.conf. > I can make perfect conversation with my friend with the only exception of > both parties being on private ip address. > There can be four situations when a call is established. > 1. A and B are not behind NATs > 2. A is behind a NAT, but B is not. > 3. A is not behind a NAT, but B is. > 4. A and B are both behind NATs (different NAT of course). > Among the four situations, 1, 2 and 3 works fine. (I guess externhost and > localnet did the trick) > But situation 4 does not work. > In situation 4: > When I call my friend, I get only one antenna bar with an exclamation mark. > (I am not talking about the iPhone's wifi bar, nor carrier's bar. I am > talking about Softphone's bar). But my friend has no problem with the > antenna bar. And both of us cannot hear anything. > When he calls me, now he gets one antenna bar and an exclamation mark, but > my antenna bar is just fine. And we still don't hear anything. > One time, there was only one time we could hear each other. He called me and > for the first 3~4 seconds, we could hear nothing. But after that we could > hear each other. I don't know how it worked. It was just one random success. > This is really weird. I tried Viber in the same situation and Viber works > just fine every time. > Could anyone give me any plausible explanation for this phenomenon? > Ed
It shouldn't matter but I would only define my actual network range for localnet. What NAT/Firewall are you using in front of the Asterisk box? I have had to turn off the NAT randomize port option or turn on the NAT static port option. I have also sometimes had to disable the SIP ALG. Ryan -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
