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Software Communications

Open Source Telephony Gives Customers Control 83

Linux.com's Tina Gasperson recently had the chance to sit down and talk with Thomas Howe, a small shop owner working to help implement open source telephony solutions. "Howe says open code is the key to highly customizable phone systems that truly meet the needs of individual companies. 'The telecom world has typically been a very closed environment. In terms of technology and deployment, they control every aspect of the experience. The idea of being open and allowing customers to have control is a radical thought.' But that is just what Howe is doing. Howe bases his custom communications solutions on Asterisk, the popular full-featured open source telephony engine that many companies are adopting as they move away from legacy phone systems in an effort to save money and gain more control over their infrastructure."
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Open Source Telephony Gives Customers Control

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  • Why does that article feel like it's an advertisement? Linux.com has a particular audience and I don't believe you need to sell them on the merits of open source software...

    It just felt empty.
  • Junction Networks (Score:5, Interesting)

    by mysqlrocks ( 783488 ) on Monday December 17, 2007 @06:51PM (#21732334) Homepage Journal
    I am a small business owner and we use Junction Networks [junctionnetworks.com] for our telephone system. Their services are standards based (e.g. SIP federation) and don't have stupid limitations like Vonage (who charge per line instead of just usage) or Skype (who don't federate with other SIP providers). The hosted PBX is pretty nice or you can use IAX trunking if you prefer to run your own Asterisk PBX. I am not affiliated with them - just a happy customer. If you're a small business I recommend you check them out.
    • I've used Junction for a few years and I agree, they do a great job. I have one account conected to Asterisk at the office and one of my lines at home is connected to a different Junction account. Most of our calls go out the office pbx, but if it was unavailable, I still have the Junction line.
    • (My first post attempt didn't make it so here goes one more try.) I too have used Junction for a couple of years and we're pleased with their service. I have one connection to our asterisk pbx at the office and one to my SIP phone at home, which has three lines. VOIP for small business can be risky but if you have people you trust (or can hire them) you'll be ok. I've also used many other providers with good results both in the USA and Europe. We do a weekly VOIP Users Conference http://voipusersconference [voipusersconference.org]
  • Sometime back, I stumbled upon this iphone+bluetooth+asterisk combo app [nerdvittles.com] (though works for any bt-phone) and was surprised by the power of asterisk.. From curiosity, How easy is it to set up an asterisk for personal purpose at home? Anyone implemented a setup and care to share some opinions?
    • How easy is it to set up an asterisk for personal purpose at home? Anyone implemented a setup and care to share some opinions?

      I've been wanting to know what's possible for some time as well. Please repost this as an 'Ask Slashdot' question.

    • Re:Power of Asterisk (Score:5, Informative)

      by Albanach ( 527650 ) on Monday December 17, 2007 @07:22PM (#21732624) Homepage
      I've been using it domestically since 2002. It has run all our home phones for three years now.

      If you want to do something quickly, there are live CDs that will have you set up in very little time. If you like to tinker, get Asterisk: The Future of Telephony from O'Reily and a linksys spa terminal adapter so you can use an ordinary phone.

      Something like the spa-2002 is nice becuase you get two lines. It's easier to experiment if you have two numbers. Linksys make WIFI dongles for these too. They're nice because you can then add a phone line anywhere in the house. Once you'r ehooked you can think about spending money on SIP phones. The SNOM phones seem to be favourites, or you could get a Cisco number like you might have in the office. They give you the nice big LCD display to play with.

      Once you have played for a little, you'll probably never look back. Remember you really do want 256kb+ upstream bandwidth and if your home network is doing anything else you'll really appreciate some QoS. You also want a stable network connection. You can use Codecs to work round bandwidth, latency 100ms tends not to be too bad, but jitter is a killer. if your pings are all over the place, you'll end up sounding like an extra from Dr Who.
      • by Scutter ( 18425 )
        I tried messing with it a few times, even with the live CD's. The biggest single hurdle I faced was simply with hardware. I couldn't get a straight answer on what to use to interface with my POTS line. There were a lot of chipset suggestions, which doesn't help when buying hardware. The only actual hardware recommendations I could find were for things that hadn't been produced or sold in years, or if they *were* still available cost several hundred (sometimes several thousand) dollars.

        Ultimately, what I
        • by Dan667 ( 564390 )
          A hardware suggestion - Digium TDM400P (1 Port FXS). Would allow you to route incoming POTS calls into your asterisk box. To be honest, I would check your internet connection if it has enough throughput and just run everything voip. (there were some other postings on how to check if you have enough) This is a pretty informative site. http://nerdvittles.com/ [nerdvittles.com]
          • by Scutter ( 18425 )
            I bought an FXS way-back-when, I believe it was that same card. I may be mistaken, it's been some time. The quality of it was horrendous (and later confirmed by numerous postings on Asterix forums).
            • by Dan667 ( 564390 )
              There are a lot of tweaks you can make to the gains, but I would agree to a certain degree. If you are looking for crystal clear phone calls you will never get it with this card, but it after messing with the gains it meets my needs. No one has every asked me if there was a bad phone line, etc.
      • I've been using it domestically since 2002. It has run all our home phones for three years now.

        My experience is similar. Many home routers allow simple QoS adjustments (priority by the port is a good one, give the phone port a high priority). I have found that lag of up to 150ms is tolerable. 250ms is doable and anything higher is awful, man on the moon lag. I've used about 10 SIP/IAX providers and a lot of different phone hardware. There is a live conference every Friday at 9 AM Pacific, 12 Noon Eastern and 17:00 UTC about VOIP and asterisk, the VOIP Users Conference that has been going on since M

    • Re:Power of Asterisk (Score:5, Informative)

      by Dan667 ( 564390 ) on Monday December 17, 2007 @07:29PM (#21732704)
      I have set up asterisk at home and love it. In fact, I wrote a voicemail app for it and put it out there for free - http://www.littlejohnconsulting.com/ari/ [littlejohnconsulting.com] My install has voicemail space till you fill up the harddrive, call attendant, and unlimited routing/call forwarding options for the lines I have. My favorite is what I have heard called the ex-girlfriend option, where you route calls that you know you do not want to never-neverland. Your don't have to know they called.

      I have it running on an old 600Mhz machine, have a digium card, and used http://freepbx.org/ [freepbx.org]. If I had it to do over again, I would not have any phone line hardware (drop the digium card) and do everything voip buying the service from a voip vendor.

      I found it to be a lot of fun and to meet my needs it did not take to much effort. Lots of help is out there now.
    • There are a couple of roll-ups that include Asterisk, a GUI, and other apps along with a Linux distro on a single CD. I personally have used trixbox for a home server with a telasip VOIP line. If you just want to do an easy home install on an old machine or VM you could start with one of these.

      Trixbox [trixbox.org] is one of the most popular, I found it very easy to install and use. However they were featured in yesterdays article about a phone-home "feature" that allowed Fonality to run code on an installed machin
      • Don't forget http://centpbx.com/ [centpbx.com]

        CentOS based with FreePBX and other goodies

        I dropped Trixbox when Fonality bought it and never looked back
      • Another one that seems to be gaining a lot of traction and backed by a lot of online help, is PBX in a Flash [pbxinaflash.net] - it is created and supported by Nerd Vittles [nerdvittles.com].

        They have loads of info on their site, including the obvious requests, like how to setup a new system quick & easy, what phones to look for, what hardware cards or peripherals to use to interface with POTS lines, as well as a list of VoIP providers that they have reviewed and recommend (or don't,) which you can read at Providers - The Best of Nerd Vit [nerdvittles.com]
    • I'd love to know how to implement nets.

      I've been kicking this around, and haven't found a good solution. I could conference everyone together, then create lists of people that create a net and whenever someone talks on that net, it whispers to the specific people -- but whisper isn't really mature in asterisk as far as I can tell.

      Doing multiple conference calls and bridging them together sounds good, but I haven't been able to find ANY documentation on doing something like that.

      I have been posting
    • Re: (Score:3, Informative)

      by Doug Neal ( 195160 )
      I tried it out, and wasn't particularly impressed.

      All the documentation seems to assume that you're using Digium's POTS cards in your Asterisk box. So does the code. Asterisk insists on using a clock in these cards as its timing source, and if you aren't using one, it needs its own Linux kernel module to provide timing (which isn't in the main kernel tree - cue lots of unnecessary messing about with compiling modules). Worse still, if you're using a 2.4 kernel, it abuses your USB controller for its timing s
      • Re: (Score:2, Informative)

        by DataSpring ( 757974 )
        I expect the timing issue is due to the fact that in the telephony world, if you have a phone switch (a Class 5 Phone switch [wikipedia.org]) and are using a TDM circuit from that switch (A T-1/DS-1, DS-3, OC-3, and so on) you must submit to the timing produced on the circuit from the phone switch...if you didn't, and produced your own timing, then inevitably, you would end up with echo. (This is the way it works in all traditional telephony.) If the device at the other end isn't providing a timing source, you must provide
  • by G4from128k ( 686170 ) on Monday December 17, 2007 @07:04PM (#21732482)
    How open is open, and how open should open be?

    Telecommunications is a critical commons and I fear what phishers/advertisers/malware distributors might be able to do it they are given too much access to the code.
  • by Tim Ward ( 514198 ) on Monday December 17, 2007 @07:20PM (#21732612) Homepage
    ... to make telecoms systems work properly. Most protocol suites are sufficiently poorly specified that a certain amount of folklore, which you can only gain from years in the industry, is necessary as well as a careful reading of the specs.

    I really do hate to think what would happen to the world's telephone system if vital pieces of infrastructure have code in them that is randomly hacked around by amateurs, well-meaning or otherwise. "Look, this must work, look, it says so here in the spec, my code follows the standards, it's all the other guys who are wrong, they should fix theirs" ... yeah, right.

    (Example: as soon as you do something to base station code which looks perfectly allowed according to the GSM specs but is out of the ordinary, ie is not something that current live systems routinely do, you start coming across "bugs" in phones ... and I mean mass market rock-solid stable phones like bog standard Nokias ... where the "bug" is its failure to implement some detail of the standard that has never been needed in a live network. Great fun to play with, to be sure, if you've got your own private base station and a bunch of test SIMs, but you don't want people doing this sort of playing in important parts of the live system.)
    • Sounds like the so-called professionals made a complete and total hash of the live system since the documentation for that system doesn't in the least match reality.
  • Let's say I wanted to start up a company with, say, 20 employees, probably growing to three times that in a year. I know how to setup a LAN, what to purchase in PCs and peripherals, what office furniture to get, and even what legal hurdles I have to negotiate in order to get the business license and set up all the HR stuff.

    What I don't know is how to set up a IP telephone system, from scratch. I've seen lots of stuff about IP Telephony in the office, and hear it is the great new thing, but every time I se
    • Here you go, though today's news from Trixbox isn't exactly comforting, you can still use it if you don't mind being watched. Oh yeah, and hopefully they don't make you use the pro version, cause then you have to configure your system through their networks... oh what a pain that is. Configuration should be done locally.

      Okay, enough ranting, and more links.

      http://www.trixbox.com/products/appliance [trixbox.com]
    • by hurfy ( 735314 )
      exact thing here

      and what internet line would be needed for 8 lines?

      and will a regular (high speed) fax machine work on them?

      Interesting timing, i am writing up instructions for installing one of our vintage (read analog that no vendor will touch) PBX in a remote office. Curious if the internet line + phones needed + learning to setup is cheaper than POTS lines and my $100 ! PBX system ;)
      • and what internet line would be needed for 8 lines?

        For G.711u (the standard codec used on traditional digital lines and defaulted to by most VoIP devices), the spec says 64kb/s but I see a flat 80kb/s per call on my routers. I have very few customers using lesser codecs, so I can't say what those use.

        and will a regular (high speed) fax machine work on them?

        Good luck. In theory it works, but even within a switched LAN I've had it regularly fail over G.711. With T.38 it's supposedly usable, but the hardware is rare. The best reliability I've seen involves dropping the speed to 9600bps or below.

        Interesting timing, i am writing up instructions for installing one of our vintage (read analog that no vendor will touch) PBX in a remote office. Curious if the internet line + phones needed + learning to setup is cheaper than POTS lines and my $100 ! PBX system ;)

        I have no idea

      • You should learn/run an openser server with Asterisk as the voicemail/POTS out. Or possibly freeswitch if you are ready to get in the steep and deep end.

        Keep the fax on a POTS line.
      • easy..set yourself up with one of the many hosted VOIP companies out there. The small office I used to work for bought Comcast (only option for broadband that we had, other than a T1) for their hosted phone system. All you have to do is buy phones for the system; usually the VOIP vendor will recommend them to you. We paid $99/month for the internet line plus $50/month for each line.

        While a PBX might have been cheaper; this was certainly far less headache.
    • Call Cisco.
    • Most business-level ISP's are hosting or at bare minimum reselling VOIP service. Let them handle it. VOIP (like POTS) is its own special terminology and way of working.
    • by lurcher ( 88082 )
      I am in exactly the same situation, setting up a IP based system for a small company.

      I would start by reading this http://www.asteriskdocs.org/ [asteriskdocs.org] it helped a lot, its biased towards asterisk, but also explains a lot of the terminology
  • I was skimming my rss reader and thought it said open-source tele-PATHY...darn...
  • Not Everything... (Score:5, Informative)

    by mikecx ( 1204636 ) on Monday December 17, 2007 @07:56PM (#21732964)
    Having spent the last few weeks setting up Asterisk and such i've found a few things major to most smaller companies that it doesn't do well. For one, SLA or Shared Line Appearance (aka, everyone can see who's on line 1, pick up the call and answer it). While the code exists it's hard to use, the documentation is poor, and the people in the IRC channel only manage to mock those who don't know that the secret lies in a pdf buried in the subversion source code and will only expend the energy to type out some cryptic code to their bot that points you at the same tired document that doesn't answer your questions. The SLA in Asterisk *works* if you have the right equipment and the time to set it up but Linksys does it one way, Aastra another, and Polycom one more. The goals of the Asterisk project seem alright but the developers seem to have it all wrong. Rather than focusing on the features users want and getting them right they are kinda hacking it all together and deciding it needs to be worked on later. Rather than making it run well on most systems, they sacrifice things to make it run on the 133mhz machine hiding in Edison's garage. I know making it light on memory is important as too much will make voice quality horrible but there comes a point when the user side of things has to be more important. My last gripe with Asterisk is that there are a few different people working on different versions all at the same time seemingly getting nowhere. Take Asterisk, elastix, FreePBX, OpenPBX, and whatever else may be out there and get all the devs to work together and get it up to something that feels mainstream and open source worthy. Asterisk is a great project but it's still got a ways to go before it's ready for massive rollouts. The only reason i'm setting one up is that the current BizFone system we have crapped out and has been crap from day one.
    • by deniable ( 76198 )
      Just curious, but why would you still put calls on line n when you have a full PABX? I just did a quick google and Asterisk appears to have call parking features. (Call parking may be what you're looking for.) How good the documentation is, is anyone's guess.

      • Re: (Score:2, Informative)

        by mikecx ( 1204636 )
        I work at a small company between 15-20 people total including interns, cleaning staff, and others. While we have 3 people that normally answer the phones sometimes they aren't available. If they can't answer the phone it doesn't make sense for the call to go unanswered while the rest of us are there. Instead of having to have all the phones ring or having to have someone run over to the phone it would be nice to have the line light up and be picked up anywhere. In a business with more than 3-4 lines I agre
        • by deniable ( 76198 )
          Fair enough. You may also want to check for a code to pick up the phone remotely. This is only useful if someone can hear the other phone ringing.
    • by NullProg ( 70833 )
      the documentation is poor, and the people in the IRC channel only manage to mock those who don't know that the secret lies in a pdf buried in the subversion source code and will only expend the energy to type out some cryptic code to their bot that points you at the same tired document that doesn't answer your questions

      I would say, in general, this is the number one problem with OSS. It's not the code quality or features, its the lack of quality documentation the users expect.

      As a programmer, I plead guil
      • by mikecx ( 1204636 )
        I agree for the most part but I think the amount of users using apache / gimp / samba and the quality of support behind those is actually really good. The documentation and config files are dense but well commented and most things make some sense. Granted .htaccess files, virtual hosts and other features may not be idiot proof but just typing .htaccess into your search engine of choice brings up plenty of help. I am also quite guilty of not writing documentation for the software I write (and work in the sa
    • True, learning Asterisk has a steep learning curve. I was part of a team implementing Asterisk, we paired the server with Cisco phones (what joy in flashing to go from SKINNY to SIP). Then the fun of figuring out little problems like echo and porting over phone numbers. This was all with fairly decent networking and Linux abilities. But the stuff works, that company has multiple offices all running on the one server. Coding every feature that they might desire is being done. I have moved on to being an IT
  • by sjbe ( 173966 ) on Monday December 17, 2007 @08:11PM (#21733070)
    We installed an Asterisk based solution at a company I owned. All open source stuff. The features and performance per $ were amazing. We loved how it could be customized to our needs and it saved a bundle. Or so we thought. Problem was, our "control" only lasted to our door. We had a great system but our ISP (Charter Cable in this instance) would drop packets, had misbehaving routers, and generally didn't give a crap that half our customers couldn't hear us or could old hear every other word we said. We just could not get a phone system that worked because we couldn't get reliable bandwidth. (yes we had uptime "guarantees" which were worth the paper they were written on since we lacked the resources to sue) Of course we could have bought their solution for about 10X the cost and were almost forced to.

    IP telephony is the wave of the future and I'm very positive on the open source stuff. But unless you have copious and reliable bandwidth, beware that you may not have the control you think you do.
    • Re: (Score:2, Insightful)

      by dumeinst ( 664891 )
      I've tried running our full pbx (15-20 users) off our comcast cable and I have nevered suffered so much in such a short period of time. We ended up going to a T1 and haven't had a single problem since. The ideal situations for us is a channelized T1 with 12 channels voice and 12 channels data. That way we can do incoming on the voice (cheap) and outgoing on the data (cheap)
    • by grasshoppa ( 657393 ) on Tuesday December 18, 2007 @12:05AM (#21734552) Homepage
      For now, you need to treat any pbx like a pbx. The intranet infrastructure you control, so you can do what you want there. But from the copper out, you need to go with what's reliable. And for now, that's copper/traditional telco lines.

      I have done several pbx installations, voip and otherwise, and let me tell you: People love asterisk. I get them setup with copper/t1s, and everything else is gravy.
      • (Disclaimer: I work at a VoIP company)

        A lot of our customers don't really have many problems with their providers ADSL, but at the end of the day there's no QOS.

        We pretty much solved this by offering our own ADSL and having QOS on their LAN, then QOS enforced at our side on the ADSL. It's pretty reliable so far, but it's only really required because some ISPs offer substandard high-latency bandwidth even on "business" DSL.

        That said, where reliability and a guranteed number of channels is needed - we always
  • Asterisk is great! OK, its configuration language is pretty sucky, but we've done some amazing things with it -- too long to post in a /. article. Just look at the slides instead: (1.1MB PDF file) [roaringpenguin.com]

  • Before I lay out my argument, let me start by saying that I really hope that one day it will be, but I am not holding my breath.

    OpenSource or any other type of software that runs on a general purpose computer will not be as efficient as say a machine from Panasonic or other phone vendor, you might be able to play with the bells and whistles more, but it will just not make the grade.

    • Phone switches are very specialized hardware. They are definitely not you average Intel box A great deal of what they do
    • by dskoll ( 99328 )

      Phone switches are very specialized hardware. They are definitely not you average Intel box

      Oh yeah? I opened my expensive NEC Electra Elite phone system and discovered it was an Intel box running (of all things) embedded MS-DOS!!!

      The software they run may very well be Linux based, but its not you average Linux.

      Our Asterisk PBX runs on bog-standard Debian Etch and it's great.

      People trying to run their phone systems over things like cable modems are just nuts, you need a dedicated T1 channel for eac

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