I know that I did not understand you =) --- On Sat, 11/21/09, Scott Lawrence <scott.lawre...@nortel.com> wrote:
> From: Scott Lawrence <scott.lawre...@nortel.com> > Subject: Re: [sipx-users] sipX Bridge > To: "Jordan Turner" <jordan.turner1...@yahoo.com> > Cc: sipx-users@list.sipfoundry.org > Date: Saturday, November 21, 2009, 4:04 AM > On Fri, 2009-11-20 at 19:39 -0800, > Jordan Turner wrote: > > > When I say "test" bridge, I mean the default > SBC. I removed the > > standard sipXbride-1 SBC and just entered a new SBC > called "test." > > That's it. Internet calling would use this > "test" SBC and calling > > between remote workers WORK just fine. If I use > the default > > sipXbridge-1, the calls between remote workers do NOT > work. With the > > above I just mentioned, I do not even have an ITSP > setup in the mix > > yet. > > > > If I use an ITSP like Flowroute as mentioned earlier, > I use the > > default SIP Trunk SBC of sipXbridge-1. All of my > remote workers can > > dial through the ITSP and make "normal phone calls, > BUT still NOT > > extension to extension or between remote workers by > extensions. for > > example, I cannot dial extension 1401 - my boss even > when we are > > connected at the same time to the sipX server remotely > (my extension > > 1402) - it works if I don't use the sipXbridge-1 from > standard SIP > > Trunk setup. > > > > I hope you can understand here, if not let me > know. I've tried to > > explain it as closely as what the label says. > Forgive my lack of > > formal VoIP experience; however, I know my FW rules > are correct as > > verified by Michael Pilcher book and pfSense blog on > it. 5060 and > > 5080 are open on the WAN that maps to the sipXecs > server's 5060 and > > 5080 ports. I've been working on this for 2-3 weeks > straight now and > > I've tried many different configurations including on > softphones - > > removed eyeBeam and using XLite now (paid to free). > > I know that you think you've described your problem, but > please take my > word for it that you have not. You've described a > series of changes and > different things that fail. What you need to do is > carefully describe > exactly one configuration and exactly what doesn't work and > exactly how > it doesn't work. > > Your problems are separable. I understand that you > think that there is > a relationship between having a SIP Trunk role configured > and not, or > one name and another, and the failure of extension to > extension > calling... this is not the case. A SIP Trunk (unless > you've done > something very odd with your dial plans) has nothing at all > to do with > extension to extension calling. > > So... just ignore the SBC and debug why you can't call > between > extensions. Most likely it is because your > registrations are not > working properly (the fact that the phones can originate > calls does > _not_ prove that they are registered). > > If the extensions that are having problems are remote, > see: > > http://sipx-wiki.calivia.com/index.php/Configuring_remote_workers_cheatsheet > > > > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org 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/
