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Communications The Internet

New Open Source VoIP PBX 151

Posted by timothy
from the tin-can-and-string-compatible dept.
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."
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New Open Source VoIP PBX

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  • by Anonymous Coward on Wednesday February 23, 2005 @07:01AM (#11753945)
    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?
    • by Anonymous Coward
      This looks kind of like SER http://www.iptel.org/ser/

      For sip only there are a few options available.

      Anyway I41 like the swiss army knife approach of asterisk, would love to see encrypted IAX2 though.
    • by Lumpy (12016) on Wednesday February 23, 2005 @07:28AM (#11754028) Homepage
      gobs of people and businesses.

      3com's voip phone systesm are this way. and you can get pots to SIP adapters for much cheaper than the specalized cards that asterisk uses.

      asterisk is a cool project, but it's huge and designed to interface to lots of legacy hardware.

      personally sip alone works great for me. I can have incoming voip calls on one of my 2 voip lines routed to different phones in the house and do other neat things that are certianly doable with asterisk but this project I was able to be up and running in 3 hours of tinkering. I NEVER was able to get asterisk working the way I wanted after 2 weeks of tinkering. Too many configuration options and features that I will not use.

      but then that is why I run thttpd and not apache for my home web servers :-)
      • by quarkoid (26884) on Wednesday February 23, 2005 @07:52AM (#11754094) Homepage
        "asterisk is a cool project, but it's huge and designed to interface to lots of legacy hardware."
        Nope. Asterisk is designed so that if you want to interface with lots of legacy hardware, it's easy to write an application interface to do so. There's a big difference.
        "and you can get pots to SIP adapters for much cheaper than the specalized cards that asterisk uses"
        Or, alternatively, you could just use those cheaper adaptors with Asterisk as we do.

        We have built our business based on Asterisk and have several SIP-only installations as well as SIP and TDM combined installations. We regularly undertake product surveys, but as yet we haven't found any product which can match Asterisk, let alone beat it!
        • can you recommend one of these systems (or another) for me? I've never done any PBX before. I am looking for a system which would let me connect approx 30 properties in a small housing association together in the UK, unlikely to be more than 50 connection points. Some of the software discussed sounds like it is for really big projects, has way too much functionality... On a definitely tight budget, hence interested to find out if it's possible to set this up as a community rather than pay for a high end con
      • What POTS to SIP adapters do you use for this?

        I've seen lots of SIP to POTS adapters for hooking traditional phones to a SIP PBX, but not the other end SIP PBX to POTS phone line.

        If I can get such an adapter that would solve my problem of not being able to track down any single line FXO cards or suitable Intel winmodems (I bought 2 and they ended up not being the ones I ordered.)

    • 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.
    • Actually not so. sipX is a fully functional PBX that supports SIP phones, and analog phones through an ATA. Support is also in development for supporting the Citel gateway, which means sipX will support Nortel digital PBX phones. the sipX family includes a PBX (SIP proxies, SIP media server, and browser-based configuration system); SIP routing proxy (same pieces, less the media server), SIP spftphone and a SIP UA
      • Actualy a gateway will be the ATA in that case and not sipX. sipX will be just a sip router and that's all. In fact i doubt that sipX can be ever called a PBX or a gateway.
        • You are certainly corect that the ATA is a gateway, just like a Digium card is a gateway, and that sipX is not a gateway, it is a Linux-based, SIP, PBX and proxy. That, by the way, is the value of the SIP architecture. It works w/ any SIP end-point - gateway, phone, etc. And in SIP, the end-points (phones, gateways, applications)are intellegent. Some featurs live in the phones, and phone can broadcast their state (aka presence) to other applications. That means every PBX feature does not need to be imp
          • I'm a developer for Yate (http://yate.null.ro) and i have try to compile that sip stack. After getting to 2 Gb i have quit. I have look for like 6 months for a good SIP stack, we've ended by doing our own SIP stack. You want to know why? Because it dosen't crush. In fact one of Yate jobs was to work as a H.323-SIP signalling proxy, because sipfoundry stack didn't manage to route more then 15 calls. What are we talking about here? About telephony which must be stable or about some Windows game?
            • thanks for clarifying. the stack actually runs at about 40 calls per second now and is completely stable. the reSIPprocate stack is also on SIPfoundry, is used for high-performance session border controllers, among other applications, and is also fast and stable. Of course, these stacks may not have been fast or light enough for your particular needs, and so you have built your own. Great. But that does not reflect on the quality and value of stacks for other applications, nor of the sip PBX, proxy, so
              • It seems that i've end up with a stack that have 90 K of code and is flexibile enough to be used for a client, or a server or a proxy in a single or multi threaded program. Is called YASS (Yet Another SIP Stack), and when is compiled it actualy have something like 83k now (of course we will develop it). I've try sipfoundary in the past and i know how huge is it. And the test i've told you about have been for a SIP - H.323 signalling proxy not for pure SIP. For pure SIP, SER which is a very good SIP router c
                • sorry, missed the question. the sipXTAPI, which includes the stack, media processing, call processing and the API is about 850k and the stack alone is about 500k. Anyway, you insist on being insulting w/out out being productive. Best of luck
    • There are plenty of people who use SIP-only setups. But it seems silly to limit one's self to just SIP. You end up having to rely on other gateways anyway to hit other networks and loose much of the powerful features that other protocals provide. My biggest beef with SIP is that there is no good provision for inter-PBX communication in the protocol. Every SIP channel is a separate voice channel. This has it's place...especially on the client end. But for communicating between PBXs, things like trunkin
    • Well, this isn't technically true or necessary anymore.

      Asterisk was a vehicle to get people to buy Digium products that interface the telephones with the PBX, and interface the PBX with the telcom lines (PRI, T1, etc.).

      Now that you can purchase SIP-to-POTS adapters for $50 and real SIP desk phones for under $70 from www.voipsupply.com [voipsupply.com] and hook them directly into your network, there really isn't a need for Asterisk anymore. We really needed and wanted a pure SIP solution.

      Asterisk wants to use its IAX pro
      • Asterisk is not the only software who can do IAX-SIP. Yate can do it also. Yate has nothing to do with Digium hardware. It support "and" Digium hardware but that's all. And IAX products are mainly at the same price as SIP. I'm not a fan nither for IAX, nither for SIP (i like H.323 :)), but you can deal with both of them quite decent.
        • Cool, my point however is that the economies of scale (lots of generic SIP hardware) brings the cost of SIP down. For better or worse the industry has decided on SIP and not IAX.

          Kris
    • > Asterisk's primary appeal is that it integrates POTS

      From what I've used of Asterisk, past 2 years, POTS is its main drawback. From buggy Digium cards to poor QA. 'cvs checkout -r HEAD' seems to be the standard tech support. Thanks but no thanks, especially when 'cvs checkout' broke it in the first place and the last 4 HEAD checkouts didn't fix it. Asterisk and POTS is NOT by any measure production quality.

      r7
  • by SizL (824056) on Wednesday February 23, 2005 @07:02AM (#11753946)
    And it's PBX4Linux. http://isdn.jolly.de/ [jolly.de]
  • by FirienFirien (857374) on Wednesday February 23, 2005 @07:02AM (#11753948) Homepage
    I think the number of acronyms per slashdot article might be an indication of its geek-tech depth...
    • by Anonymous Coward
      What is GTD supposed to mean?
    • You like Acronyms then telco is the right tech for you, they were doing acronyms when others were still trying to invent a language. Rumor is the original 26 letters are a secret telecom acronym refering to the name of the original Network Architect. I've got the 15th edition of Newtons Telecom Dictionary - from around 2000, last I checked they were publishing about once every 8 months to keep up with Telecom terms. (course the bubble did burst so they may have slowed since then)
      aybabtu - didn't make it in
  • Here's more of'em (Score:4, Informative)

    by Anonymous Coward on Wednesday February 23, 2005 @07:27AM (#11754026)

    The only? What is this, IDG?

    I can think of at least two right away:

    • SER
    • Yxa

    There are probably others, feel free to add...


    • SER is not a PBX, it's a proxy server. A proxy server is a component of a SIP architecture and you would almost certainly (but not absolutely) need one INSIDE a PBX.

      SER is a fantastic little proxy though -- just not a PBX.

      A PBX includes media processing, voice mail and other 'enterprise' features.
  • So, Is this really an important story?
    • by Bothari (34939)
      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...
      • There a number of commercial vendors who sell, configure, and support Asterisk; this is a good option for people who aren't up to doing the configuration themselves, and no worse than if they had gone with a commercial-only product.
      • by bastion_xx (233612) on Wednesday February 23, 2005 @10:15AM (#11754825)
        Try asterisk@home [sourceforge.net] for a good distro that should do most of the easy stuff "out of the box".
        • That would be great if all I wanted to do is run Asterisk. Unfortunately, I'm not made of money so I only have one server and have to run all of my services on that. I've successfully got Asterisk working for my SIP service already, but I would have loved a simple, near-pre-rolled distro of it back when I was installing it... something I can just install on my existing Linux box and have it work.

          Where's that?

      • Wow. Actually that's exactly what Asterisk is perfect for: small company (5-50 users) that wants ringing phones, voicemail, etc. I've implemented it in that capacity many times. Of course it can do much much more but you don't have to use those features--they're just *there* when you do the install. BTW, the install really isn't too tough--my first install took a couple days to get right and I was a newbie to Linux. Now I can install and configure Asterisk for a small office in a couple hours max.
  • by Alistair Cunningham (20266) on Wednesday February 23, 2005 @07:53AM (#11754098)

    What's particularly interesting with this product is that it includes a VoiceXML browser.

    For those who aren't aware, VoiceXML [integrics.com] is a cross platform markup language, visually similar to HTML, for writing IVR [integrics.com] 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 [integrics.com], 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.

    • From the sipX site [sipfoundry.org]:

      sipXvxml - VoiceXML processing engine

      The sipXvxml project combined with OpenVXI v2.0, sipXportLib, sipXmediaLib, and sipXtackLib produces the SIP voicexml engine used to power the sipXpbx's project's voicemail and autoattendant features.

      License

      sipXvxml is distributed under the Lesser General Public License (LGPL).

      Documentation

      Coming soon!

      Riiiiiiiiiggghhhhtttt. Cute, guys. Wake me up when you've written some fucking docs.

  • The Real Issue (Score:4, Insightful)

    by osewa77 (603622) <naijasmsNO@SPAMgmail.com> on Wednesday February 23, 2005 @08:04AM (#11754125) Homepage
    Is that another VoIP company decided that Open Source is a good strategy. That's the real story!
    • Sometimes it is, sometimes it isn't. We looked into deploying GNU's oSIP [gnu.org] library. Unfortunately, it wasn't up to snuff with what we needed (namely, four VoIP lines on a device), and the amount of rearchitecting required could not be justified in terms of man-hours and time constraints. So it's an off-the-shelf SIP stack for us. Given a longer deadline and several people working on the problem, it could have been done, but the realities of business meant that in our case open source was simply not a good str
  • Jargon Buster (Score:5, Informative)

    by Jack Taylor (829836) on Wednesday February 23, 2005 @08:06AM (#11754131)
    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.
    • Thanks!

      Sip for me was something one did with wine until this useful definition.
    • Re:Jargon Buster (Score:3, Informative)

      by CounterZer0 (199086)
      Actually, SIP is 'Session Initiation Protocol', as specified in RFC 3621 (http://www.ietf.org/rfc/rfc3261.txt)
    • 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.

      Being the last option certainly helps consistency, but fast delivery? Come on, the CowboyNeal extension is excessively bloated. Just check out this artist's conceptual visua [cmdrtaco.net]

  • copper 2 pair. I remember when telco technology was simpler. The early telephone networks were really not much more then advance tin-can and string communication. Humans manually switching and routing calls. Alexander Graham Bell likely had no idea of what he was creating with his invention.
    What I love about telco technology is how much of it is piggybacked onto the original design - the basic phone - mic/speaker isn't dramatically different now, copper 2 pair is still used - we've piggybacked technolog
  • And can any of these systems let me make POTS phone calls for the price of setting one up and a broadband connection?
  • I'm still unsure of why people would setup their phone systems this way. I looked at setting up a VOIP box in my house, but it seemed to me that you had to pay a company for the phone service routing. The prices weren't cheap either. Am I missing something?
    • $7.99 a month for the number, plus $0.02 per minute. With Asterisk, you can keep your land line (for the directv) and have it route local calls via the landline for the unlimited free local calling your teenagers need.
    • The prices weren't cheap either. Am I missing something?

      Yeah, like .02 cents a minute US and extremely cheap international calls. I have asterisk running two remote offices in my company, and the price and performance vs. the Cisco Call manager I booted out are more than worth the headache of asterisk. And admittedly, asterisk is a huge pain in the rump, but managing it keeps me 733+
  • by tburt11 (517910) on Wednesday February 23, 2005 @11:20AM (#11755419)
    I recently deployed asterisk in a few locations. It was admitedly tough. I was unfamiliar with the world of telephony, and new to VOIP. I read lots of wiki pages, and read through the extremely detailed configuration files, and with some trial and error, I now have a fully fledged PBX in my home, and my workplace.

    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!

  • by aminorex (141494) on Wednesday February 23, 2005 @11:42AM (#11755628) Homepage Journal
    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.
  • For front ends, there is switchvox [switchvox.com], which wraps Asterisk.
  • I see that sipX is in fact just a LGPL'ed sipxchange (pingtel corp.). sipxchange is quite possibly one of the messiest software products ever to have been created. Please look another way :)
  • How do I hook up a local phone line to asterisk? I want to use my local phone line for all local calls and VoIP for international calls.

    So how do I link my local phone line cord to a box running asterisk? do I need a special card or adapter? How much are these and are they compatible with Linux? Please suggest one if you can.
    • Asterisk is Linux only (I believe there is a windows port out there) so you don't have to worry about that. Figure about $100 for a card to interface the analog telephone with Asterisk. You can also get a FXS to SIP interface (Sipura makes one) but I don't recall the price off hand.

      Your Asterisk setup should be on a relatively dedicated system (Mine is also my home web server) not on your desktop machine.

      For more info, see http://www.voip-info.org

      Good luck!

      Jeremy

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