Peter, Did you look at http://www.cudatel.com? Probably just what you are looking for. GUI goodness based on FS.
SDR Peter J. Zandvoort wrote: > Matthew, > > I'm about in the same boat as you are, just on a smaller scale. We have a > ton of Nortel telephony gear, but it's time to move out of the 90's and > enter this millennium. My Cisco quote was in the same ballpark as yours. > > The Cisco stuff is mature, rock solid, meshes very well with their network > gear and is actually relatively easy to set up and maintain if you know your > way around IOS. I just refuse to pay that kind of money for yet another > semi-proprietary solution. > > After looking at various asterisk distributions, SipX, 3CX and > what-have-you, I've come to the conclusion that FreeSWITCH is by far the > most advanced platform out there. Its architecture and performance is > literally light years ahead of the rest and I have yet to come up with > something that it can't do. But all that comes at a price: The learning > curve is like scaling a brick wall. The developers and the community are > great and available, but just starting out with SIP and voip in general, > this may not be the best platform. So let the blasphemy begin :) > > SipX was a breeze to install (insert CD, boot, next next next...) and looks > pretty solid. I believe they actually use FreeSWITCH for their voicemail and > conferencing, internally. I just couldn't get my head around their GUI, ACD > was too basic and had all kinds of issues getting stuff to "just work". > > 3CX (Windows Only) was completely painless. It just worked. But I'm still > not convinced that I want to run all my voice on a single windows box. Plus > it's not free/open/etc and I don't want to lock myself in again. > > Although it's an asterisk based solution, I found trixbox to be very easy. > Setup is automatic and everything "just worked". The GUI is simple and > logical enough that I can let somebody else handle the day-to-day phone > setup and basic admin. I have my doubts about it scaling to 250 users, > though. > > This may be a completely flawed strategy and I may very well be shooting > myself in the foot by doing this, but I plan on piloting a trixbox install > with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH > box next to it for the more advanced stuff. Once I get more comfortable with > the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, > I have a feeling that that trixbox is going to get phased out... > > Peter > > > -----Original Message----- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of > mkitchin.pub...@gmail.com > Sent: Tuesday, November 03, 2009 11:10 PM > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones > > Michael Collins wrote: > >> On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com >> <mailto:mkitchin.pub...@gmail.com> <mkitchin.pub...@gmail.com >> <mailto:mkitchin.pub...@gmail.com>> wrote: >> >> I'm working on an alternative to a $120,000 Cisco phone system that my >> >> company is looking at. I got Freeswitch installed on CentOS last week >> using the Quick and Dirty instructions. That part was painless. We >> had a >> few 7940s laying around. After some wrestling with it, I got the >> latest >> SIP firmware installed and what I hoped was a functional config >> (attached). X-Lite phones can call each other no problem. 7940s >> can call >> X-Lite no problem. Anytime I try and call a 7940, it goes straight to >> voicemail. I attached a log file that shows the activity when >> trying to >> call a7940 from X-Lite. >> X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is >> nshplpbx1.unix/10.85.0.53 <http://10.85.0.53>. Everything is on >> the same LAN. Different >> subnets, but no firewalls. >> I didn't see anything that said posting attachments was frowned >> upon. I >> apologize if it isn't appropriate. I'm guessing this is something >> simple >> and I'm just clueless on how to diagnose the issue. >> I'm not tied to using this model for good, but it is what we had >> laying >> around. Any help would be greatly appreciated. Next step is >> configuring >> it to talk to Verizon VOIP over a DS3. >> >> Thanks, >> Matthew Kitchin >> >> >> Matthew, >> Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We >> think you'll find FS is as powerful as any software out there right now. >> >> Here's a handy wiki page that will help you get the diagnosing skills >> you need: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> I'd say first thing to do is capture the SIP traffic to see if there >> are any clues. A "normal temporary failure" doesn't give you a lot of >> detail. :) If you're new to SIP debugging then the best thing to do is >> to capture the SIP trace and put it in the pastebin. >> (http://pastebin.freeswitch.org) >> >> You can also join the IRC channel #freeswitch on irc.freenode.net >> <http://irc.freenode.net> and get some real-time help. There are some >> sharp folks in there, not the least of which are the three main >> FreeSWITCH developers. >> >> -MC >> > Thank you. I think I did what you are looking for. I stopped FS and > launched this command. > TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch > and captured all output to http://pastebin.freeswitch.org/10965 > Does this tell you anything? > I'm definitely new to SIP and phone system admin in general. I have > plenty of network and Linux experience. With that in mind, someone on > this mailing list emailed me directly and said SipX would be a better > fit for me. Is that blasphemy for me to even mention? I went through the > documentation and the provisioning aspect and web interface do look > tempting to a novice. I apologize if this is like trying to buy a chevy > at a ford dealership. I'm looking to deploy about 150 handsets at a > corporate office and then 10 to 12 handsets at 120 remote locations. We > are moving from an old key system, so our current features are very > limited. We just need a few ACD groups, call history, and the other > general basics. I first found Asterisk and read about some of the > shortcomings. FS looks like the most robust solution. I have no idea > where SipX would fit in. The people here are obviously a very > knowledgeable group and I would gladly accept any thoughts, comments, etc. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
