shouldn't it be easier to deploy since there is not dual NAT (nat on server end, and nat on client end) to worry about? the server has a public IP accessible and I just blocked all unnecessary ports. I believe the no audio or one way issue is attributed to the server being behind NAT and the phones behind another NAT.
Any workaround I can do? I need to get this setup by tonight :) On Mar 8, 2010, at 5:06 PM, Tony Graziano wrote: > On Mon, Mar 8, 2010 at 4:46 PM, Francis Tinio <[email protected]> wrote: > Hi > > I deployed another sipx server (server is remote located with public IP > (firewall is within server, no NAT). I provisioned 3 polycom phones. All 3 > phones are created via phone group, so all have the same settings. All 3 > phones have successfully registered with the server as well and all 3 phones > can make outgoing calls. > > However, incoming call does not work. Only the last extension provisioned > can receive incoming calls. With the other 2 extensions, I get a user is > unavailable and get directed to voicemail. Even internal calls within > extensions, the 2 phones cannot be reached and i'm getting the same message > that the user is offline. > > What could be wrong? I checked the phone settings of the three phones and > they are all similar. All 3 phones are registered and can make outbound. > > Please advice. > > Thanks > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > I have never been able to get sipx working properly in the open supporting > remote users and trunking. At the same time I am not sure if the design ever > really took that into account. > > http://list.sipfoundry.org/archive/sipx-dev/msg22401.html > > >
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