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 <m...@freeswitch.org> wrote: > Tom, > Welcome! Sadly, your experience is not unique... > > On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom > <paveraw...@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@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_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400
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