New Open Source VoIP PBX
dsginter writes "It looks like Asterisk isn't the only open source PBX game in town anymore. sipX, as the name implies, is a SIP-only PBX project released under the LGPL. A noteworthy feature is the inclusion of an out-of-the-box web-based management console. Read more about the release over at Voxilla."
SIPx appears to be a PBX only, with no way to attach real phones. Asterisk's primary appeal is that it integrates POTS and SIP. Who uses SIP and SIP alone?
And it's PBX4Linux. http://isdn.jolly.de/
Perhaps they were all reading the article? *gasp*
I know, it's more likely the comments were intercepted en-route by a pack of marauding ducks, but hey, it could happen.
I think the number of acronyms per slashdot article might be an indication of its geek-tech depth...
Browsing with +2 to insightful posts and a higher threshold makes the average post seen seem a lot more ingenious
PABX: Private Automatic Branch Exchange
or
Private Access Branch Exchange (less common)
*buzz* its Session Initiation Protocol
thanks for playing. =)
The only? What is this, IDG?
I can think of at least two right away:
There are probably others, feel free to add...
What's particularly interesting with this product is that it includes a VoiceXML browser.
For those who aren't aware, VoiceXML is a cross platform markup language, visually similar to HTML, for writing IVR applications. VoiceXML pages can be served from any web server, and converted to voice on an VoiceXML browser. It interfaces seamlessly to Text To Speech and Voice Recognition servers.
My company, Integrics Ltd, does Asterisk, Cisco Call Manager, and SER installations. Up to now, we've done IVRs using Asterisk AGI for smaller systems, and VoiceXML on Cisco 2800 routers for larger systems. Being able to run VoiceXML on a free platform on Linux is going to be very interesting our customers. Needless to say, we're getting up to speed on sipX, and will be offering installation and development services as soon as it's mature.
Is that another VoIP company decided that Open Source is a good strategy. That's the real story!
There seems to be some confusion over the acronyms on this topic, so I thought I would clarify some of them:
PBX: Private Branch Exchange - this is basically a computerised telephone switchboard, allowing even fairly small organisations to manage their own telephone networks at low cost.
SIP: Session Initiated Protocol - this is the protocol that is standard on most voice-over-IP devices.
COWBOYNEAL: Circulation Of Worthless Broadcasts Over Your Nearest External Authentication Location - this is a special extension to the voice-over-IP standard allowing fast delivery of esoteric technological news to compliant devices. It also has the convenient property of always being last on selection fields in the user interface.
One good turn - gets all the covers.
Crucially important. Asterisk is the pits to implement for 80%+ of the situations where a open source voip pbx would be useful.
Don't get me wrong, it's amazingly powerful and does just about anything except wash windows... as long as you can get it working properly. But it's not the right tool for a small (think 5-50 people) company which only wants a simple PBX to connect their phones...
Enjoyed your post, but should point out that only one pair is needed for telephone communication, including ringing. The two pair is a more recent wiring standard.
Yate (Yet Another Telephony Engine) is also a gateway and a PBX.
It supports H323 (much better then asterisk), SIP (with a nice stack that it can be actualy reused), IAX2 (with a forked version of libiax2), and ISDN (PRI and BRI) using zaptel drivers.
The best part is that is much more flexibile then any other similar project around. Is not like sipX just SIP based, and is not like Asterisk a emulation of PSTN over VoIP. Is a real VoIP server that actualy deal also with PSTN.
Try asterisk@home for a good distro that should do most of the easy stuff "out of the box".
I have never sought out a GUI interface for asterisk.
If I wanted a GUI interface, I would have looked for a MS based solution. Isn't that obvious?
From what I have read, and experienced, IAX is a superior protocol to SIP, principally due to it's handling of NAT and firewall issues. It just works, and it works well. I can send an IAX adapter to the far side of the world, and have the user plug it in. Without the need to add rules to their router, I can connect and Voila, they are talking.
I am very pleased with Asterisk. I have only begun to utilize it's vast capabilitites.
It appears that SIPX is targeting the user who wants simplicity. Most windows users are attracted to simplicity. Ergo: Asterisk is like linux, manually configured and extremely powerful. Sipx is like windows, give me a dialog box to type in my phone number, and that is all I want.
DISCLAIMER: I have never used SIPX, but a quick look at the website, and pulling up blank pages for the readme's tells me alot!
If only sipX would support IAX2 protocol, we'd have
a really useful component which would peer with
Asterisk servers and be operable over stupid NAT
devices such as the majority of connected systems
use to connect to the Internet.
-I like my women like I like my tea: green-
"Avoid employing unlucky people - throw half of the pile of CVs in the bin without reading them." -- David Brent