Raul,
 
I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what 
you describe. When I was trying to procure funds for a FreeSWITCH solution, I 
looked for the same information you're after, but came up with little. I'll 
briefly describe what we're trying to accomplish, and the tools I'm using to do 
it. This is probably more information than what you are looking for, but maybe 
it will also benefit someone else.
 
We had several schools with aging or dying PBX's or KSU's. Each site had 
something different system, and was supported by a different VAR. Of course, 
the VAR's charged some outlandish fee to make onsite repair visits. Some number 
of Centrex lines supplied each school's dial tone. All in all, we had a very 
outdated and financially draining mess. Our district's long term goal had been 
to move to a more unified phone system. That made sense for many reasons, the 
chief of which was cost. We already had a strong fiber WAN in place. Why not 
use that for trunking and eliminate the monthly cost of the Centrex lines? 
That's the path we started down.
 
Being a public entity, we had to be sure to explore all possible avenues. We 
looked at everything from traditional PBX's with IP add-on modules for trunking 
to a full blown Cisco CallManager solution. With third party proprietary 
systems, we were just never able to find the sweet spot between required 
feature set and cost. Would Cisco have been a workable solution? Absolutely. 
Could our small, rural, K12 public school district afford that? Not in a 
million years. I looked at several software packages -- some open source, some 
not -- but always came back to FreeSWITCH. The scalability and active 
development community were major factors for us.
 
Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For 
hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz 
processors, and 4GB of RAM. I have mirrored disks set up with enough space to 
accommodate users' voicemail. Each server will average only about 60 voicemail 
boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We 
have always been a Novell shop, so SLES is naturally our Linux distribution of 
choice. We chose to go with server hardware at each site so that in the event 
of a WAN outage, we would still at least have intra-building and emergency 
communication (see below).
 
Telephony Hardware. Each of our servers includes Sangoma hardware. We actually 
looked at doing IP trunking to a carrier from our network core, but decided to 
use telco provided PRI's instead. Presently, we have two PRI's that connect to 
a FreeSWITCH server at the center of our network via a Sangoma A102 dual port 
telephony card. All calls to and from the PSTN traverse this primary server. 
Servers at each remote site include one of Sangoma's A200 analog cards. 
Emergency calls to 911 route out over this analog card through one of at least 
two POTS lines that remain connected at each site. Not only does this provide 
some redundancy in the event of a WAN outage, but it ensures proper caller 
location is delivered to the 911 dispatcher. Granted, there are some other 
solutions for the latter, but this seemed to be the most cost effective 
solution for us.
 
Telephone Desksets. We chose to go with Aastra for the telephones. The standard 
phone that we will place in each classroom and office is the 9143i. This is an 
attractive phone with an adequate feature set at a price we can afford. The 
person that is primarily responsible for answering the phone at each site will 
have an Aastra 57i and some number of 560M expansion modules. We have purchased 
roughly 300 Aastra desksets.
 
Logical Layout. As new sites come online, their primary phone number is being 
moved from the Centrex to our PRI group. All inbound calls hit our primary 
server, and then FreeSWITCH bridges to the appropriate secondary server based 
on the DID it received. On the reverse, each servers dial plan is set up to 
route outbound calls (save 911) to the primary server where FreeSWITCH bridges 
with Openzap. Site to site calls, accomplished via four digit dialing, do not 
hit the primary server. Outbound calls to the PSTN deliver the site's DID as 
the calling number. In other words, if a user from site two calls my cell 
phone, I see site two's published telephone number on my caller ID. Our dial 
plans are set up so that receptionists at each site still answer all outside 
calls. If not answered, the call fails over to an IVR. Should we ever decide to 
do so, we are now perfectly positioned to have all inbound calls to the 
district answered by one operator or IVR. "Welcome, and thank you for calling 
Avery County Schools."
 
Stumbling Blocks. Problems we've faced so far have primarily surrounded Openzap 
and the Sangoma Wanpipe driver. FreeSWITCH developers won't mind telling you 
that this is an area that is currently not well "funded" and not 100% complete. 
There is some known issue where voice channels on the PRI get stuck in the 
wrong state and become unusable. We have experienced this a couple of times and 
have not been able to make or receive calls. Bouncing the Wanpipe driver has 
fixed this each time. We have also had trouble with DTMF detection across the 
PRI. If a user hits the IVR, it is oftentimes difficult to get it to properly 
recognize the digits that are being keyed in by the caller. This can be very, 
very frustrating to a caller that doesn't want to deal with an IVR anyway. The 
developers have suggested to me that this is a problem with the Sangoma's echo 
cancellation goofing up Openzap's ability to interpret the DTMF. The Sangoma 
hardware does have its own DTMF decoder and API, but the Openzap code currently 
does not make use of it. I have created a patch that makes use of the hardware 
decoder. We have been running it in production for a couple of weeks, and that 
does seem to have helped the problem. The problem hasn't gone away altogether. 
Those have been our two biggest issues, but we haven't let them hold us up.
 
Conclusion. Of the five sites that will be on this system, one is fully 
functional with calls inbound and outbound from the PSTN. A second site is up 
and running with full outbound PSTN access. Their inbound DID is scheduled to 
move over to the PRI in one week. The server has been worked up for a third 
site, and the phones are starting to roll out. Sites four and five should come 
online by the end of April. Currently, I don't have numbers compiled for things 
like concurrent calls. At this point in my project, it is just not important. I 
really don't think our implementation will ever push FreeSWITCH's abilities in 
that regard. I base that statement primarily on other users' benchmarks, and 
what I've heard some are doing in carrier class environments.
 
FreeSWITCH has made our project viable. An open source solution was the only 
way we could meet all of the project goals and stay within our budget. 
FreeSWITCH has proven to have all the features we require in a district wide 
phone system. It has not locked us into annual support contracts with third 
party vendors. I could go on with the accolades. However, I'll end this 
terribly lengthy post by saying that, overall, we have been very pleased with 
our choice to go with FreeSWITCH.
 
The information in this email will seem very elementary to most people on this 
list, but having a message of this nature in hand would have made me feel much 
more confident the first time I ever went to my supervisor to mention something 
called FreeSWITCH.  :-)   Thanks Tony, Brian, and Mike for a great product!
 
 
Ben Holtsclaw
Network Engineer
Avery County Schools
Ph:  828.733.3567 x2301

>>> On 2/18/2009 at 11:13 PM, Raul Fragoso <r...@etellicom.com> wrote:
Thanks guys, this is very useful information. 

Anyone else willing to share your experience ?

Regards,

Raul

On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote:
> Hi Raul,
> 
> In my company (http://www.globant.com) we're using FreeSWITCH for high
> quality conference services, integrated with OpenSIPS
> (http://www.opensips.org) and Asterisk. Its performance is pretty
> good.
> 
> Pablo
> 
> On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang <red.rain.se...@gmail.com> wrote:
> > bandwidth.com has a service called phonebooth which is developed upon
> > freeswitch.
> >
> >
> > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso <r...@etellicom.com> wrote:
> >>
> >> Hello FreeSWITCHERS,
> >>
> >> My company is currently creating a suite of applications which uses
> >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a
> >> prospect to have our first customer installation - a governmental
> >> department. That is a tender to have an IP-PBX installation to connect
> >> their four office branches, each one with about 300 users - which I am
> >> sure FreeSWITCH is able to handle. Since this is an official tender,
> >> it's part of their protocol to ask about real sites using the product.
> >>
> >> Having said that, would you mind sharing some information about your
> >> experience with FreeSWITCH deployments ?
> >>
> >> No need to give many details, but a short summary with company name (if
> >> possible), when it was deployed, server equipment, number of users,
> >> number of concurrent calls, what kind of functions and services are used
> >> and overall capacity of the system.
> >>
> >> I would really appreciate if you can share that information. And if you
> >> guys agree (and explicitly manifest your agreement), I can compile the
> >> information in the FreeSWITCH wiki under a "Use Cases" page so it can
> >> serve as a common reference as well.
> >>
> >> Kind regards,
> >>
> >> Raul Fragoso
> >>
> >>
> >> _______________________________________________
> >> 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 
> >
> >
> >
> > --
> > Henry Huang
> > UniC Solution - Communication Unified
> > VoIP & Open Source software Consultant
> >

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