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/
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