Cuney, I don't think just 'opening" ports at either end would work. If you are passing out over the open internet and have a firewall, it either needs to be sip aware and configured properly, or you need to be able to pass the traffic over a vpn and be able to resolve the DNS SRV records accordingly.
Tony >>> Cuneyt M <[email protected]> 02/04/09 1:27 PM >>> Hi Michael, I actually follow the tutorial (and verified SRV records resolved ateach end) and setup the two end-points. Used the full domain in the gateway address field (also tried with IPaddresses, using domain name proved to be working v.s. IP) What happens is, when i dial from site A and site B phone rings thenhangs up. When i dial again from site A to B, B rings and I manage to answer.After about 12 seconds it hangs up. Ports 5060 and 5090 (both tcp+udp) are opened on site A and B's router. Does this behavior ring any bells or a pattern? Offtopic: I've had continuous freezing after i update 3.8 to 3.10.3 andfirmware 2.2.2 to 3.1.1 (rom 4.0 to 4.1.1, even tried 4.1.0). I have tried the combinations that are suggested in the list but thefreezing is still happening. As i suspected of a possible corruption in xml configuration, i'veexported the user+phones, deleted everything under TFTPROOT folder andimport back the user+phones and the Polycom 330 files and deployed theconfigs after editing Phones under a single Phone Group (i.e. set thedate time to 24 hour and disable most of the Features as it doesntapply to 330). Now i'm still running 3.10.3 and switched the firmware back to 2.2.2and ROM 4.0 What i suspect is, the polycom template files under /etc/sipxpbx/polycom/mac-address.d now changed to 3.1.(1) config XMLs. And if it did i don't know howto reverse it back to 2.0 XMLs. Do you have any knowledge on this or how to verify and/or switch backto correct XML templates (i am assuming these files under mac-address.dupdated with the Yum sipx 3.10.3 update...) Any help would be greatly appreciated. Cheers. Today's Topics: 1. Re: multi-site sipXecs deployment (Picher, Michael) Subject: Re: [sipx-users] multi-site sipXecs deployment From: "Picher, Michael" <[email protected]> <o:shapedefaults v:ext="edit" spidmax="1026" /> <![endif]--> <o:shapelayout v:ext="edit"> <o:idmap v:ext="edit" data="1" /> </o:shapelayout><![endif]--> <a moz-do-not-send="true" href="http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs";>http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs<o:p></o:p> <o:p> </o:p> The big thing is to get the two dns domains sothey canresolve each other’s addresses…<o:p></o:p> <o:p> </o:p> Mike<o:p></o:p> <o:p> </o:p> mailto:[email protected]] On Behalf Of RobertJoly Sent: Wednesday, February 04, 2009 9:34 AM To: [email protected] Subject: [sipx-users] multi-site sipXecs deployment<o:p></o:p> <o:p> </o:p> Hi, Ihave atwo-site setup with a sipXecs at each site and I want to connect thetwo sitestogether so that a user at a given site can call a user at the othersite. I was looking through the sipXecs wiki to figure out the bestpractices for deploying such a configuration but didn't find anything. Ifsomeone has a pointer to such information, could you please send it tome?<o:p></o:p> Thankyou. bob <o:p></o:p> http://list.sipfoundry.org/mailman/listinfo/sipx-users </[email protected]>
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
