Since Digium/Mark refuses to include any code that isn't copyrighted Digium/Mark, I'm afraid quite a few developers may be effectively excluded. By requiring a 'giveaway' to Digium/Mark, we stand the chance that the project one day will change its license to a (more) closed license (as just happened to MySQL, for instance). We already have channel drivers not included in the asterisk CVS because of this (chan_oh323 and chan_capi).
I am not saying this license change will happen, but there's always a chance, and for some the requirement of giviing away the copyright to their own code may be a hinder to write it in the first place.
roy
On Saturday, Nov 8, 2003, at 10:16 Europe/Oslo, WipeOut wrote:
Can I add to this and say that another thing that could be hindering the takup is "Single System" VoIP scalability and a certain amout of "Enterprise" flexibility..
Let me explain those two..
Before you start reading these and thinking "This guy is mad!!" let me just say that I love Asterisk and use it every day, but if M$ and IBM are getting into the game there is cause for concern.. The features I am going to talk about are very much "Future Dreams" becasue to impliment them would probably mean re-creating the entire code base from the groud up so I don't expest to see the features ant time soon.. I do think that these sorts of featured will be in the IBM and M$ IP PBX's and that is why I think Asterisk needs them..
So lets get started..
I know that many Asterisk servers can be connected together to scale the size of the system but this is still a problem because it is a headache to manage.. What is needed to get the big enterprise players on board is the ability to manage the PBX as a single entity no matter how many "servers" there are.. "servers" should simply be add on modules to the overall PBX to improve its VoIP call volume handling power.. I think the only way to achive this would be to make Asterisk a "clustered" software that sits a level above the "servers".. The VoIP phones will see one "Asterisk Server" that listens on a single IP address per subnet on the network but behind that single system image could be one, two or fifty servers providing the processing power for all the calls, and as power is needed you simply have to add servers.. If you need more PRI lines just add a Digium card to a server and enable that server as a "gateway" node in the cluster..
With in this model the voicepath between the "servers" in the cluster needs to be dynamic so the shortest path is always used (IAX can probably handle this quite well already), and CDR must be accurate maybe one or two of the nodes needs to allocated the task of being the CDR server and all other servers will feed back to the central server with the call logging information..
In "Enterprise" flexibility I am taking about user and phone management and services..
On the phone management side (and I know many don't seem to like the idea) but a platform independent full featured management interface is needed.. If its done in Java or web based running on the Asterrisk sever itself, similar to how webmin has its own web server, does not matter but we live in a world now where admins like GUI management tools..
Leading on from that is an "Operator Interface" for receptionists and phone operators to be able to manage calls.. See which lines are busy, connect calls and the various other things that these interfaces do..
Next a monitoring interface (somthing similar MRTG would probably do it..) showing server loads and statistics so system management and upgrading is easy to see and plan for..
Then the need to support hot desking.. By this I mean that the phone and the user need to be seperate entities on the system.. then the user can sit down at any phone on any desk run through a login procedure (either on the phone or in some easily accesible interface) and all their calls will then be routed to that phone.. I know there are hacks and work arounds to getting this kind of functionality using queues and the Asterisk DB and various other options but it needs to be a standard working system..
Finally an automatic provisioning system.. New user joins the company, click a button on the management interface and give them their extension number and extension password.. no editing files and restarting servers or anything like that its all done behind the scene..
So did I just thumb suck these concepst out of this air?? not totally..
Last year I did a contract at a large comnpany in London and was working on a user provisioning system.. This company has thousands of users in a single building (and a single PBX) in London, and thousands more accross the country.. It was a provisioning system so I needed to talk to the telecoms guys to see if we could automatically provision the phone extensions from the central application.. So a lot of my ideas here come from what I saw they had and things they said they would like to have..
Anyway I will stop rambling on now..
I still think Asterisk is great for SOHO and medium businesses, and when the Digium multiport analog or a BRI card (I know ISDN cards can be used but it would be nice to have one that provided Zaptel timing and one that would probably be a lot cheaper than the current active ISDN options.) comes out it will be great for the small companies as well..
Later..
John Todd wrote:
Yes, it is a well-kept secret, which is a shame since it obviously fits so many different requirements. Here are some late-night musings as to why new users coming to Asterisk is only a stream when it should be a river:
1) No >1.0 release. In fact, no release structure at all really. (Hold your flames: I know this is to be remedied soon, along with backtrack patches for security/stability.)
2) No books (yet.) This also is going to be remedied soon.
3) Advocates fall (generally) into two camps:
a) IT staff who have much more on their minds than being VoIP advocates, and who normally are told what to do. Even if they have experience with * in testbed situations, the larger vendors come in and throw whitepapers/jargon/FUD at executive staff, who make telephony decisions, thus overruling clueful staff.
b) CLEC or other telephony-oriented people who will try very hard to prevent anyone from knowing what they use, or how they use it, since that is a competitive disadvantage if others should start to use the same software-driven architectures. There are some obvious exceptions to this, but you'll very rarely see (ever?) any posts by the two or three major IPCSP's that use Asterisk as part of their core systems.
There are of course others who do not fall into one of these two camps, and those are the people being the "zealots" getting conversions to Asterisk. Personally, as an example, I have over two dozen institutions, companies, and very clueful individuals that I've introduced to Asterisk simply based on chatting with them. (excluding clients, who already have intentions on installing Asterisk.) The time it takes to explain why Asterisk is so useful is quite labor-intensive, actually, and the educational process takes some time even with the most clueful engineering types, simply because there are so MANY things to take into consideration with Asterisk and any telephony questions in general.
4) Hardware vendors are still blowing enough "QOS" issues around that it obscures open-source VoIP solutions. "VoIP won't work" is still a claim I hear EVERY DAY, until I disagree and tell that person that I'm disagreeing with them over a VoIP call that crosses a continent twice, across the public Internet (and three carriers.) This is obviously not Asterisk-specific, but it's certainly an issue that scares people away from OSS solutions that don't include "magic hardware."
5) I would say that it's becoming less of a secret, so don't give up hope. The almost-unmanageable flood of newbie posts to the Asterisk lists in the last two months or so is evidence that success is sometimes more of a headache than one would want.
In short, nothing in the above 4 "worry" items scares me, and Asterisk is and will become the telephony platform of choice for a large percentage of conversions to VoIP in the coming years. Fret not: you'll be the apache of VoIP soon enough.
JT
Asterisk has got to be about the best kept secret in telephony. I've seen
numerous articles on slashdot about VoIP, even in relation to Linux and
only *once* has the post even mentioned Asterisk. Am I missing something,
or is Asterisk clearly a good potential player in any kind of linux-based
soft-switch idea?
Mark
On Sat, 8 Nov 2003, Dave Cotton wrote:
For those who don't wake up at 5.00 am and start reading /.
http://searchnetworking.techtarget.com/originalContent/ 0,289142,sid7_gci935769,00.html
--
> Dave Cotton <[EMAIL PROTECTED]>
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
