Jordan, Great news. I think you have the "ITSP cheatsheet", if you don't just google it, you will find it on the sipXecs site. Follow it step by step, literally. Your ITSP will come up and work for you.
The ITSP is no mystery, and Ranga has written up some nice tips that will help you get it up and running in now time. It involves turning on NAT traversal, configuring a new Gateway, which will route through the SipXbridge, and then it will have you set up your dial plan. Just follow the cheatsheet to the letter. Good Luck to you. BTW, ITSP's come in all shapes, colors, and quality of service. If you are trying to wow your company, be sure to have trunks that are reliable and work well, or you will have a disaster. That industry is going through some consolidation now, and some of the smaller firms will not survive I suspect. There have already been some to shut down suddenly, leaving customers stranded to port their numbers in too short a time to get it done. -----Original Message----- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jordan Turner Sent: Friday, November 20, 2009 11:37 PM To: Scott Lawrence Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] sipX Bridge You are correct Scott, thank you. After re-reading your comments, I tried differently. I know that my main issue is not knowing the exact order of configuration. I messed up the remote workers simply because I chose to tie the default SBC to Internet calling when it had no reason to be - discovered from your key explanation. I have gotten ITSP and remote workers working via incorrect order / procedures - but never together at the same time. At this point, I am looking for proper next step into adding an ITSP like Flowroute - not HOW but WHAT order? Add another SBC, then a Gateway for Flowroute to connect to, the a dial-plan? Do I create a new dial plan or custom? One can re-read these docs, and even book which I purchased, but until you start driving, it does not completely click or make sense. At the end of this, I plan on having a video of how all this works for the next newb! Thanks again. --- 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/ _______________________________________________ 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/
