Digium Blog Appears to Censor Comments
This is pretty sad. I hope they are just short-staffed.
I read this article yesterday and was surprised to see that buried in a defensive stance against sipXecs and Nortel were comments claiming my experiences with Asterisk where "rhetoric" and calling us a "Competing Open Source Project" (I am not sure what we are competing for, I thought we were on the same side, but ok)
http://blogs.digium.com/2008/08/19/asterisk-the-global-leader-in-open-source-ip-pbx-and-telephony/
Here's what he said that caught my eye:
"You should recognize that the Asterisk rhetoric from the sipXecs and FreeSwitch teams refers to Asterisk over 4 years and several versions ago when they last looked at the feature set."
This guy clearly didn't work there when I was an Asterisk developer and he has no idea the last time I looked at Asterisk because I still have had patches added to Asterisk SVN as recently as Nov 2006. I did a heck of a lot more than "look at the feature set" thank you very much.
Anyway,
I posted a comment with my feedback and almost 24 hours later it's still awaiting moderation. The sad thing was all it said was that we should all work together.
Here is my comment anyway since they won't post it:
I didn't know open source projects had to compete. They are all free aren't they? I think that's the whole problem here. Calling my list of valid issues with Asterisk rhetoric, won't make them go away. Pretending Asterisk does not suffer from any problems and only pointing out it's strengths is not the way to make it better.
Please admit that I have done more than most people are willing to do for completely FREE to try to make Asterisk better for several years. I only know enough to itemize the issues from that *long* experience as a an Asterisk developer.
I, in fact, invented the whole idea of the "function variables" that now are rampant in Asterisk 1.6 and there are *plenty* more things I could list if I wanted to. I also see plenty of ideas we have already implemented in FreeSWITCH starting to crop up in 1.6 as well. This is the nature of open source. If Digium chooses to actually cooperate with the open source telephony community there is much to be gained for all.
" Competition is a fact of life and good for users as it draws the best out of all of our efforts. "
Asterisk was seen as Cisco and Nortel killer app back then (people started to migrate all their proprietary solutions to *). Now in 2008. (five plus years since Asterisk was born), Digium is going after FreeSWITCH and other open source projects to compete with? Isn't that frustrating for Digium people?. I thought in 2004. that VoIP equals open source and Asterisk. I predicted soon end of Call Manager and similar products. As we all see now, I was wrong.
Digium never responded to less than few problems inside the core (hence the FreeSWITCH and Callweaver). Nobody from Digium ever bother to answer non-biased technical objections such as one that Anthony wrote about on reasons to start FreeSWITCH in the first place. Instead, we hear stories in 2008. similar to ones from Microsoft marketdroids. That's just pure spreading FUD.
As a user of both FreeSWITCH and Asterisk, I have to emphasize FreeSWITCH community for their support and fast reaction to users' wishes, suggestions and problems. It is a friendly community where people actually DO get responded back by dev team. With Asterisk this is no case for years now. I still get e-mails about bugs going back from version 1.2.
To conclude: marketing stories such as 1900 cps won't suffice with people anymore, cause Asterisk has de facto nowadays a bad reputation. I urge Digium to actually go back to square one, and start dealing with real problems, and not making them look smaller. There's only one Microsoft.
Note: i'm using asterisk 1.4 but i honestly think 1.2 has more stable and i will never move to 1.6 since its not suitable for production and i have too many custom patch to 1.4 that moving to 1.6 will require months.
My asterisk experience is 4+ years old. My freeswitch experience is very limited at the moment ( http://wiki.freeswitch.org/wiki/User:Agx )
Well, REASONS::
- if your internet is down all the blocking DNS queries will freeze people using the daemon; i see FS uses async DNS queries;
- bug reports are ignored or they try to slow down you to hell with burocracy even if you provide full debug info as required; i see FS fixes problems in 24 hours;
- patches are mostly ignored, i had to create an external packages called agx-ast-addons to support some extra functionality in 1.4 (obiovusly 1.6 cannot be used in production); i see FS included my patches in 48 hours without asking me to sign any extra legal stuff;
- codec negotiation cannot be controlled; i see FS offer different methods of controlling codec negotiation, it seem you can use G.729 MOH without having any license just implementing a logic that exclude any transcoding;
- analog card does not recognize hangup in Italy, i still have a patch for it that have always been ignored by digium (they only included the debug part of it); + BRI does not work well in Italy in PTMP mode; i see FS does not support BRI aswell but i'm going to use an external gateway in any case now;
- CDR is inaccurate, when using mysql i cannot change the Source and Target number to match what is really happening (1.6 includes the features i want); i see FS has many flexible way of handling CDR;
- Manager does not scale (1.6 fix this); i see FS has many different way of integrating with other apps, indeed i've a separate daemon that communicate directly with FS instead of using a fixed dialplan;
- lack of TCP and TLS and G.722 support; i see FS includes this;
- IRC #asterisk is manily talk about photos and other O.T. stuffs; only way to have a bit of technical excanche is probably accessing asterisk-users mailing list;
- BFL subscriptions sometimes gives problems and you require patches to do call pickup of other ringing extensions; i see FS has greater better support for Snom phones that we mainly use;
- Echo problem, choppy MOH, PCI Latency, kernel timing, ZtDummy... AAAAARGH!!! I see FS can free me from this;
Missing FS features i'm thinking to working on:
- T.38 (well i don't need it i uses a Linksys SPA to use them directly);
- Queue/Agent (can be implemented using an external daemon, nice to develop a separate call center solution);
So this is the idea of what i got about the asterisk and freeswitch project:
- asterisk: half-open source code indeed the project seems to move in the direction the company that controls him moves to instead what the community demans; poorly source code, lots of bug but also lots of interesting features; supports for Analog/PRI card and some crap support for BRI; stable versions are like beta version and beta version are like experimental code;
- freeswitch: open source code but with heavy review of the quality of the source, perhaps 3 people behind him its a bit too few but perhaps its just because there aren't yet many critical core developers; high quality implementation and design, quickly fix of bugs; lacks support for BRI cards and need more test about OpenZap stuffs; stable versions are really stables;
That's the opinion i built myself from direct work 20/24 hours everyday.
AMEN! :)
A note, the BRI support is not complete, but we have already merged the initial support for PTMP BRI.
Indeed i talked on IRC with Bkw that pointed me to the guy doing the work.
I'm checking with Xorcom people (that seems to mantain a decent version of Bristuff) to merge mISDN hfc_multi.c inside qozap.c in order to have all the 4BRI cards supported under Zaptel then OpenZap.
I'm starting with Digium 4 BRI card now and then doing Beronet, OpenVox, etc. Since i've all of them.
Anthony, we'd never censor your comments. I replied earlier tonight. Sorry it took a few days, this week I fell behind on 3 comments yours was one due to other project commitments. Please see response if you and your readers are interested on:
http://blogs.digium.com/2008/08/19/asterisk-the-global-leader-in-open-so...
Thanks again for taking the time to read and post.
...bill Miller
So glad for your efforts! Been waiting and watching for many years, first with bayonne, then with asterisk, hoping to see a fit for us. Finally, it seems, FS presents a clear direction without inherent roadblocks. Please press on with the good work and stay positive. Good tech stands on its own.
One open source project bashing another is very rare, and I think it is counter to the spirit of the community. We can, and should all work together.
Rather than bickering, cross-project collaboration will be a better use of time, but that's just my opinion.
Comment censorship is not really a bad thing. We'll all do it at some point. However, requiring registration in order to post a comment is not such a hot idea. But that - again - is just my opinion.
This seems to be the way of things at digium... "EVERYTHING" in asterisk is their own unique idea. I'm a bit surprised they haven't tried to take credit for the entire SIP spec yet (may happen soon though).
When asked at cluecon how many of the new features in asterisk 1.6 were directly related to FreeSWITCH's feature set, their spokesperson said that none of them are. I find this to be pretty amazing since there have been patches for TLS and SRTP around for a whiiiiile that weren't deemed "worthwhile" until after FreeSWITCH came out of the gate with them. Now, all of a sudden, the decision is made to add TLS/SRTP to the asterisk core.... but yeah, that must be a mere coincidence also.
mv hussein /jail && rm -f /bin/laden
yep... they seem to have ripped out the soft-based conference idea from freeswitch too, and a lot of other things... but anyway.
freeswitch will always be better ;)
...to intralanman,
The features in 1.6....come from all over the world and have nothing to do with FreeSWITCH, as our spokesperson was correct at ClueCon. The logs for 1.6 have been around for some time (approx 18 months) and all were done by various people without knowledge of (yes, coincidence) the FreeSWITCH features. It just so happens these are pretty basic features that were required (so basic that TLS/SRTP were going into my 3Com systems before I left 2+ years ago). There is no correlation to FreeSWITCH. I can also say Digium does *not* take credit for "Everything" *in* Asterisk. Our mantra is "EVERYTHING" Asterisk; we do have coding guidelines and most new code committed goes through design reviews and code quality and security reviews and is then integrated and tested in order to benefit adopters/users/developers. It must be working. Between Digium and other Asterisk-based distros there are millions of downloads and hundreds of commercial products, many in mission critical environments.
As one other commenter on this forum suggested, open source projects should be working to take over the world, not kill each other. It is unfortunate that there have been issues in the past with people and philosophies that caused disagreements in direction. It is clear there were some good and solid contributors who left the project.
If you have time, read "Soul of a New Machine" which is a book about Data General splitting off from Digital Equipment about 40 years ago where two projects inside the same company went in different directions and ended up competing head-to-head (PDP vs. Eclipse). Competition is a fact of life and good for users as it draws the best out of all of our efforts.
....bill miller
It's not important where the ideas came from. Like I said, I myself probably helped draft that list of 1.6 features where Asterisk was still in the 1.0 stage. I am proud to admit that FreeSWITCH has had a huge influence from my previous experience with Asterisk. In fact, many ideas in FreeSWITCH are based on the collaboration of all my friends in the VoIP world and I want to thank them all for their involvement.
Let's just keep swimming......
-Dory
BTW, here is my unaltered document from the day Asterisk 1.0 was released circa 2004. This document was shown on the projector with my vision of the future of Asterisk because I could not attend. Some of them now exist and some do not and many are in FreeSWITCH today ;-)
http://www.freeswitch.org/brainstorm.txt