Open Source Telephony
glengyron writes "Every Phreakers dream of Open Telephony took another step forward today when a small Australian company released their OpenPBX product at CeBIT Australia. Suddenly a massive community of programmers will be able to easily write their own Telephony applications in Perl!"
Disclaimer: I work for a computer telephony company.
That said, their product is very very poor. Almost no documentation in code (which is a MUST when you go open source), very buggy, doesn't work with most middleware (like Genesys, CT-Connect, TSAPI, Symposium etc.).
Those who can, do. Those who can't, consult.
The rest of this is going to be totally off topic (besides the fact that I am from the same place as the product mentioned in the story).
Google Cache Links
The poster was smart enough to put in google cache links to the story. Prevents a slashdotting of the original site (although google might get shirty). How about slashcode automatically include google cache links? It's easy to do. Check out merkac dot for an implementation.
Subscribers as early mirror makers
I've noticed on the last couple of stories which where savagely slashdotted, that the subscribers had a chance to mirror the articles before it became available to the rest of us (the unsubscribed rabble). On one of them (it might've been the missing matter in the universe one), an early peruse of the comments showed only 3 comments at a threshold at 2 or greater. And each of these was a subscriber (probably?) posting a mirror site.
So not only are these people paying for the privilege of seeing the stories early, they're doing work for slashdot by making sure the stories are mirrored correctly (and karma whoring quite nicely at the same time).
Maybe some official mirroring technique is called for. Not by slashdot (since they've said quite plainly that they won't mirror anything), but if there was a nice bit of code to auto-mirror every article's URL to a free web mirror (or some site which has the guts to take a slashpede).
That's all.
Cool, but useless.
* hey, it's slashdot anyway, euh, sorry, its slashdot any way.
#include "coucou.h"
get over it. dont like /. and its methods? move on. whining like a little bitch boy isnt going to do a damn thing.
-- botsex is {grep;touch;strip;unzip;head;mount}
That's cool if you want to limit yourself to H323. But should you also want SIP and MGCP support and be able to use most relevant voice codecs out there, the solution is Asterisk.
Has not been historically maintained.
"The PC required can be low end. Typical free OSs run very well on much less powerful machines than required by modern closed source OSs. Use low cost commodity hardware rather than leading edge. A monitor is generally not required for IVR servers powered by free OSs, as they can be remote administered via telnet."
So the $200 saving in hardware could quite easily be realized.Bayonne is by far the most mature telephony server out there, having been in existence as ACS long before even asterisk was a twinkle in kram's eye. Best of all, it's hardware independent, unlike either asterisk or openpbx. It supports just about every CTI card line that supports linux. They just added both PBX support and H.323 support. SIP is not far off, and if you want to be able to script complex applications in no time flat, optionally dropping out to perl, python, or php for logic, Bayonne is unparalleled. Add to that direct mysql, postgres, support, text to speech support, internatilalization, and so on and so on...
David Sugar and I actually did a lot of the preliminary work to get the OpenSwitch cards working under Linux via Bayonne. The last time I checked the best support for the OpenSwitch cards was still Bayonne - in fact, ctserver's protocol is based on the Bayonne state machine. Although you have to write the telephony parts in Bayonne's scripting language ccScript, Bayonne actually supports Perl scripts for many other functions via a gateway mechanism (TGI).
Asterisk also supports Perl scripts via their own gateway protocol, AGI. Unlike Bayonne, you can control the telephony engine from AGI to do things like dial a number, transfer to voicemail, etc. I talked to Mark Spencer, the main Asterisk developer, a while back about merging AGI and TGI into a common gateway protocol since they're so similar, but I stopped doing work for the company behind Bayonne (Open Source Telecom) recently for various reasons and the project died in my email queue.
None of the free telephony solutions have good documentation, including Asterisk, and there's no incentive to make it good since all of the companies involved like support revenue. This isn't a problem with the open solutions specifically, but rather with the telephony industry in general. Usually the best thing to do is to pick a vendor based on your project's criteria and put up with the installation process, then threaten anyone who touches it with eternal damnation. Unfortunately, with the shift towards host processing and the associated microprocessor industry economics, it's more and more difficult to get support even 5 years after product introduction, so this tactic doesn't work as well as it used to for HSP based telephony systems (which includes basically all the open solutions). If you really need it to work, you're basically stuck paying AT&T or another large firm out the nose for an expensive, proprietary, hardware based solution.
-- thalakan
The point is, Voicetronix is a kick ass company that offers good support and put has put a lot of effort into open source projects over the years, of which this is just one. In addition to projects they themselves have initiated, they have contributed to speex, bayonne, and openh323. The feature set will, inevitably, expand.