[Freeswitch-users] FreeSwitch the Right Tool?

Mon Dec 3 11:52:35 EST 2007

Yikes: I certainly never intended to be offensive.  By cavalier, I meant 
dismissive in tone.  It wasn't based on anything other than the note 
which read,

"We promised to stop making new modules or features but.......we lied."

Now, there are two things that could be going on: one is that your users 
are crying out for stability, and you are being dismissive.  Otherwise, 
it's just a joke, and your users understand your motivation etc.  All I 
was doing was trying to figure out which we were jumping into. As is my 
nature, I am generally up-front to a fault.

Again, I had no intention to offend anyone, and thank you for your response.

Bill

Anthony Minessale wrote:
> Sir,
>
> I am not sure I understand your comment about my announcement
> of the additional features being "cavalier" My understanding
> of the term, which I double checked at webster.com, is that you are 
> suggesting that I am dismissing an important matter or have some
> sort of snobbish attitude.  Quite frankly, I had decided against
> adding any new features before the release and then I decided that
> I must stick with my original plan which was to create a voicemail
> module as an exercise to ensure all the components were in place
> for a production system.  I appreciate your feedback and comments but
> the cavalier remark is hard to get over.....
>
> To answer your other questions that I did not find offensive:
>
> The reason we use external tools is because computer programing
> is complicated and takes many thousands of hours to get right.
> Many of the packages we rely on have their own independent  mission
> to provide the functionality in question and it's naive and against
> the nature of open source to bypass these tools in favor of a home
> grown solution.  The main application code of FreeSWITCH contains
> nearly 130,000 lines of code by itself and that is without counting
> any of the dependency libraries. A prime example is SIP, the Sofia
> project works day and night for years on nothing but SIP.  SIP is
> just 1 aspect of our application and we could never match the
> functionality of their sip stack on our own with all the other things
> we have to also focus on.  Our mod_sofia is about 8,000 lines of
> code on top of that existing sofia stack!
>
> The reason there are so many options is because of human kind.
> Due to interoperability, innovation and general bright ideas brought
> forth by our human race, we are forced to be flexible because one
> man's feature is another man's bug and we must allow the 2 men to
> co-exist by giving them opposing configurations.
>
> In terms of general usability, You will find, if you investigate
> closer, that the important behavior is centralized and what you
> may be interpreting as "many ways" to do things is simply the method
> of "using" these central behaviors akin to  a "skin" on a gui 
> application.  Regardless of using an embedded language, a socket
> connection or a simple dialplan, every case results in the same
> core application code doing the actual work which is intentionally
> designed in a pyramid of complexity with the top layer being the
> smallest and simplest.
>
> To summarize,  This is a very complicated application and it
> takes a lot of comprehension of telephony and multimedia to fully
> understand how it works.  Even having the experience, one will
> have to investigate the software first hand to fully understand how
> it works.  Try to remember the first time you tried photoshop!
>
> As a policy, I make no attempts to convince anyone to use my software.
> Please also consider sipX, CallWeaver, Bayonne or YATE or any other
> open source telephony application that may be available to you.
>
>
>  
>
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
> ----- Original Message ----
> From: Bill Binko <bill.binko at mapshine.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sunday, December 2, 2007 4:59:15 PM
> Subject: [Freeswitch-users] FreeSwitch the Right Tool?
>
> Hi everyone,
>
> I'm looking at FreeSwitch as a platform for an area where we're 
> expanding our business.  I was wondering if I could get your guidance 
> on whether it's the Right Tool for what I need.  I realize this is 
> potentially a Troll, and I apologize, but this seems so far to be a 
> small, positive group and I thought I'd just come out and ask.
>
> First some background.  I have several clients who want various 
> value-added phone services.  Basically, they want IVR systems with 
> some bells and whistles such as recording calls, integration with 
> their websites, and "follow me" services.  My company has done similar 
> work for clients on the GIS (mapping) side, and I have personally 
> worked with Asterisk as a true smart PBX solution, so we are 
> considering adding this to our offerings.
>
> We have a SIP-based VOIP provider in the colocation site we use that 
> has attractive pricing and we have good direct connectivity to them, 
> so we would like to this to be a SIP-only solution (no TDM hardware, 
> etc.).
>
> As we started looking at doing this with Asterisk, we found a few 
> things.  First, there are some real issues getting clear calls through 
> Asterisk when there is no TDM hardware in place. This seems to be due 
> to some timing issues, and seems to be worsened by some RTC changes on 
> Linux 2.6.x.  I wasn't surprised that there were some issues, but we 
> have had a very hard time getting it to run well on a clean distro 
> such as CentOS without custom kernel compiles, etc.
>
> Perhaps most concerning, we aren't seeing the kind of community 
> support we found (and came to rely on) in the open-source GIS space.  
> Posts to the asterisk-users board get lost on the way to the list, and 
> when they do make it through, they get responses like "buy a Digium 
> card just for timing".  Well, in our 1U servers with only a 
> PCI-Express slot, that's going to be a trick.
>
> So, I quite literally went to Google and searched for "Asterisk 
> alternative" and FreeSwitch came up all over the place.  I installed 
> it on our server and had it running in about an hour with only minimal 
> pain.  And the sound quality (so far) has been very good when just 
> playing recorded sounds.
>
> Here's my concerns about it so far - let me know if they're unfounded.
>
> 1) It seems to be reliant on a huge collection of external tools.  I'm 
> ok with running the FreeSwitch recommended versions of all of them, 
> but isn't going to be a bit fragile as these tools mature separately?
> 2) It seems young (as a project, not the folks on it).  Things like 
> this post to the homepage worry me: http://www.freeswitch.org/node/103 
> as it looks a bit cavalier.
> 3) It seems to have more options and less guidance that I expected.  
> This may be a first-impression thing, but there are many ways to do 
> everything (Dial Plan, Integration languages, etc.).  That's sometimes 
> good (Perl took that approach for years) and sometimes awful (the 
> competing opinions get in the way).
>
> We can contribute back to this project, both with 
> development/debugging/etc. time and with documentation help, and right 
> now it *feels* like the right choice.  However, I thought I'd ask for 
> some help getting past the concerns above before we jumped in.
>
> Thanks in advance for your time,
>
> Bill Binko
> bill.binko at mapshine.com
>
>
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