Hello Do no forget the rtp ports 10000 to 20000
Regards On 1/4/07, Facundo Barrera - GMail <[EMAIL PROTECTED]> wrote:
007/1/4, Bob Chiodini <[EMAIL PROTECTED]>: > Facundo Barrera - GMail wrote: > > Hi list: > > This is my first post and first off all i want to wish a good > > year for everone! well my problem is; i already installed asterisk on > > a server and created a channel and a couple of extensions, all seems > > to work just fine, y can make calls and receive them, i'm using the > > x-lite client that also works very good, this is the topology of the > > net > > > > > > (LAN - some clients) --------|| Internal interface-private IP(server > > Running Asterisk)external interface-public IP ||---------INTERNET > > > > Well i configure * to bind all address, so it's service listen on the > > two interfaces, when i make a call from a client inside my LAN to a > > client on the INTERNET, the person receives the call and listen me > > perfectly, but i can't listen any audio from him, i read about the > > issue and it seems to be a problem of nating, keep in mind that this > > server is masquerading all my LAN ips, so i can share my internet > > conenction, so when i receive a call form the outside world in fact > > x-lite shows me that the call originate from my inside interface IP of > > the server, but this is the strange thing the packets that originate > > the call from the outside world arrive just fine but when i answer the > > call i can't hear any audio at all. > > > > Any ideas how to solute this? hope not receive too much flames of this > > common issue > > > > Thanks a lot > > > > > In your SIP configs specify that the extensions are natted: > > nat=yes > externhost=<External IP address> > localnet=<Local IP subnet>/<local subnet mask> > > These are global settings. > > It might also be helpful to set canreinvite=no for each extension. > > There are probably firewall tricks you can do as well, but its early and > I'm a couple cups of coffee shy. > > Bob... > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Thanks for the answer, will try that, but keep in mind that my server don't have an static public address, i use a dynamic DNS to resolve my sip domain. Thanks a lot -- _________________________ Facundo Agustin Barrera -------------------------------------- www.openlabs.com.ar "Let the penguins do the work" --------------------------------------------- Buenos Aires - Argentina _________________________ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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