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.
Asterisk and features in 1.6
" 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.
BRI support
A note, the BRI support is not complete, but we have already merged the initial support for PTMP BRI.
par for the course
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
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 ;)