[Freeswitch-users] Hello, and stuff.

Anthony Minessale anthony.minessale at gmail.com
Fri Aug 28 12:11:04 PDT 2009


That's essentially the story of why I wrote FS.

On Fri, Aug 28, 2009 at 1:54 PM, Michael Collins <msc at freeswitch.org> wrote:

> Tom,
> Welcome! Sadly, your experience is not unique...
> On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>wrote:
>>  As a background, I ran an asterisk consulting company for about 3 years
>> that I gave up on 2 years ago after repeatedly failing to achieve any sort
>> of stability on any sort install over about 30 phones, I gave up.
> The consensus I've seen is that the larger the install, the more likely one
> is to have inexplicable issues.
>> Maybe that was wrong, I am open to the possibility that I just didn't know
>> enough and I was building things wrong, but I worked inside the asterisk
>> code (which I feel is a hopeless mess), I implemented a few small custom
>> features, anyway...
> Any software that openly admits that a function is "pure nastiness" but
> doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
> leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
> see it.)
>> I'm coming back into the VoIP space now, and I'm wondering what sort of
>> issues can I expect in trying to pick up and learn freeswitch?  From what
>> I've read on the website, it appears to have a much more sane architecture.
>> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
>> stable and worst general call quality was asterisk.  I constantly contended
>> with strange call quality issues in asterisk, lots of echo (even with
>> hardware echo cancellation cards), lots of jitter, lots of call break up
>> (even on small systems with 10-20 users, using QoS on the network, and in
>> general doing everything I could to prioritize voice over anything else).
> Again, your experience isn't unique...
>> When I used Cisco call manager and broadsoft, the voice quality issues
>> were basically non-existant, as long as the network was running QoS echo,
>> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
>> does freeswitch show a marked improvement over asterisk in this department?
>> As long as you configure QoS and have hardware echo cancellation does it
>> actually work reliably?
> We receive lots of reports that FreeSWITCH is a vast improvement over not
> only Asterisk but proprietary solutions as well. The FS architecture is, as
> you mentioned, not insane. It is well thought out and therefore highly
> flexible, extensible, and scalable. I'm not aware of anything - OSS or
> proprietary - that can match FS in these three areas.
>> Thanks for any additional information about freeswitch you can provide as
>> well.  I am a software developer primarily by trade, but I do lots of
>> consulting type work in the SME space and I've had a couple projects thrown
>> to me that require some integration with a phone system, and I just can't in
>> good conscience recommend asterisk anymore.
> Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
> development but the power users are quite happy with doing the emacs (or
> vim) shuffle with the XML config files. Furthermore, the ways that FS allows
> you to connect and control are fantastic: mod_xml_curl for dynamic
> configurations, event-socket for external control (think of it like AMI not
> sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
> the list is impressive.
> I can honestly say that every week we get new people looking at FreeSWITCH
> and saying, "Wow, this is incredible." I can definitely, in good conscience,
> recommend you investigate FS more deeply. I'm confident you'll be happy with
> the return on your investment.
> Hope it all works out for you! Join us in #freeswitch on irc.freenode.netif you want to chat in real-time.
> -Michael
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Anthony Minessale II

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