[Freeswitch-users] Hello, and stuff.

[Anthony Minessale]

http://www.freeswitch.org/node/117

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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090828/dd624ef5/attachment-0002.html