Thank you also Tony for your persistence and patience. The last comment you made about the order of the procedure should be stated somewhere. I understand that it is all stated in the longer documentation; however, it does help to reaffirm for people who have never worked with sipX or even VoIP in general. By stating it in a simple high-level overview, it help with the learning process. My problem was that I was trying to do too much and was getting "lost." So lost that I sometimes forgot the order or was not too sure of the ordering. Great community here, thanks.
--- On Sat, 11/21/09, Tony Graziano <tgrazi...@myitdepartment.net> wrote: > From: Tony Graziano <tgrazi...@myitdepartment.net> > Subject: Re: [sipx-users] sipX Bridge > To: "Jordan Turner" <jordan.turner1...@yahoo.com> > Cc: "Scott Lawrence" <scott.lawre...@nortel.com>, > sipx-users@list.sipfoundry.org > Date: Saturday, November 21, 2009, 12:46 PM > You will need to answer some things before > you begin. > Does FLOWROUTE have the ability to send you > calls on port 5080? Do they require registration or do they > send you calls by IP address instead? Does your firewall in > front of sipx have a static IP address? > > You will also need to understand what number > format is expected from you at FLOWROUTE. > For example: my itsp requires all calls sent to > them to be in the same format. +1xxxyyyzzzz (+1 and 10 > digits). I crafted my gateway to add the +, and my dialplans > make sure the rest is 1xxxyyyzzzz. > > You should follow the wiki page for setting up > the ITSP. Read through it and understand it > first. > http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration > > Also, your questions are answered here as to > whats created and in what order. SBC is created, and then > the gateway using the SBC as its route, then you would > modify/activate your dial plans. > > > On Sat, Nov 21, 2009 at 2:37 AM, Jordan Turner <jordan.turner1...@yahoo.com> > wrote: > > 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/ > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > > > > _______________________________________________ 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/
