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Open Source Telephony Gives Customers Control

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."

83 comments

  1. Advertising...? by aesiamun · · Score: 1

    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.

    1. Re:Advertising...? by dynomitejj · · Score: 0

      It seems like a lot of their articles are advertisements. What's the price for getting one of my articles on there promoting my small consulting business ? They should just come out and publish the rates.

  2. Junction Networks by mysqlrocks · · Score: 4, Interesting

    I am a small business owner and we use Junction Networks 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.

    1. Re:Junction Networks by Anonymous Coward · · Score: 0

      Sooo... You say you work for Junction Networks?

    2. Re:Junction Networks by randulo · · Score: 1

      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.

    3. Re:Junction Networks by randulo · · Score: 1

      (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.org/ live every Friday at 9 AM Pacific, 12 Noon Eastern Time. Anyone is welcome to join the conference to ask questions or share experiences.

  3. Power of Asterisk by dotpavan · · Score: 1

    Sometime back, I stumbled upon this iphone+bluetooth+asterisk combo app (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?

    1. Re:Power of Asterisk by Krishnoid · · Score: 1
      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.

    2. Re:Power of Asterisk by Albanach · · Score: 4, Informative

      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.

    3. Re:Power of Asterisk by Dan667 · · Score: 5, Informative

      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/ 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/. 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.

    4. Re:Power of Asterisk by Anonymous Coward · · Score: 0

      I'd love to hear if there is an inexpensive way to use Asterisk for a typical home situation -- I've got an analog line coming in from the phone company, and several analog lines hooked up to it. It seems like the base cost to put a card into a PC to handle even a single incoming line and a single existing phone is awfully high.

    5. Re:Power of Asterisk by kaiser423 · · Score: 1

      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 in their forums, etc. but have no good answers on how to implement something like that -- effectively a mission control architecture.

      I'm going to try and roll my own, but there's a serious dearth in this area of people who want something similar (mission commander net, countdown net, and subsystem nets that the user can choose to monitor multiple/any combination they desire).

    6. Re:Power of Asterisk by Scutter · · Score: 1

      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 wanted to do was a home PBX, running inbound calls over POTS and outbound over VOIP. I found it to be WAY more trouble than it was worth. I don't know if anyone's come out with a newbie-friendly guide to go along with the live CD's since I originally tried it, but if not, it would be REALLY NICE.

      --

      "Tell me doctor, with all of your defenses, are there any provisions for an attack by killer bees?"
    7. Re:Power of Asterisk by Dan667 · · Score: 1

      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/

    8. Re:Power of Asterisk by Scutter · · Score: 1

      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).

      --

      "Tell me doctor, with all of your defenses, are there any provisions for an attack by killer bees?"
    9. Re:Power of Asterisk by Dan667 · · Score: 1

      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.

    10. Re:Power of Asterisk by Anonymous Coward · · Score: 0

      It looks like the Digium TDM400P runs over $300 - pretty expensive to my way of thinking for *part* of a home phone system.

    11. Re:Power of Asterisk by Doug+Neal · · Score: 2, Informative

      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 source. Why they didn't use POSIX high-res timers is a mystery. Worse still, none of these ridiculous hacks even work properly - sound playback from the Asterisk box is very prone to choppiness.

      The SIP code is a complete mess, and not even anywhere near RFC compliant. Little bugs and annoyances pop up all over the place that are quite telling about the quality of the underlying code.

      CallWeaver is a fork of Asterisk that aims to fix a lot of the problems although I have reservations about how active the project is. They seem to have at least addressed the timing issue though.

      So in summary; Asterisk - nice idea, too many bugs.

    12. Re:Power of Asterisk by Albanach · · Score: 1

      Much cheaper solution is a Linksys SPA3102 - you can get that for <$80 and get 1FXS 1FXO + Wifi. As others have pointed out, the call quality tends not to be as good as you get from just plugging a phone line in or calls that are exclusively VoIP

    13. Re:Power of Asterisk by DataSpring · · Score: 2, Informative

      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) 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 one, so a timing "generator" (loosely defined - read more about Digital Channels and Time Division Multiplexing (TDM)) is included in most equipment that acts as a digital device that would serve a connection to an endpoint (A T-1 to a PBX, for instance)

      In any case, that is why the Asterisk code relies on the timing from their PRI/T-1 cards, since you *must* do so to avoid echo, if the other end is providing the timing source. If it isn't, Asterisk still relies on that clock, if it exists, since it will produce the timing for any circuits connected to it. If no such card exists, Asterisk tries to get the timing from some other source - a special module in the kernel, or the USB stack, trying to derive some source of timing that will remain reliable and steady over time.

      I'm not saying that there aren't other ways to do this, but coming from a telephony background, it was a decent decision at the time. There are also external timing sources used in a lot of telephony devices, and I'm certain Digium could find, or has found, a way to make use of them as well. This is all dependent on the fact that you'll be interfacing with the PSTN. If you are working with pure VoIP, the timing in this sense is irrelevant, and you need to coordinate other things (such as jitter, QoS, packet delay, and bandwidth) and you shouldn't need a timing source. Apparently, Asterisk still needs this timing source due to the structure of it's code.

    14. Re:Power of Asterisk by Victor_0x53h · · Score: 1

      I'm far from being a Linux guru, and I learned it in a couple of weeks. Download, and print out "Asterisk: The Future of Telephony", get a couple of cheap Aastra SIP phones (or even just one, and a soft phone, as I did), and an optional Digium card if you plan to interface with POTS (you could go entirely SIP as someone else mentioned). It was a lot of fun to make such quick progress! I later went on and installed a 5 phone voip system for the family business. Voicemail (email voicemail messages even), music on hold, and a weather/holiday closing announcement.

    15. Re:Power of Asterisk by randulo · · Score: 1

      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 March 2007. For more on that, see http://voipusersconference.org/ or http://food4wine.ning.com/ and join live any time. The conference is like a big international users group. Asterisk and the rest of open source VOIP technology is brilliant, but it does require some commitment to getting it to do what you want/need.
  4. Re:Frost dring dring by Anonymous Coward · · Score: 0

    lawlz

  5. Does "open" include the ability to spoof caller-ID by G4from128k · · Score: 2, Interesting

    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.

    --
    Two wrongs don't make a right, but three lefts do.
  6. Re:Does "open" include the ability to spoof caller by Bryansix · · Score: 2, Informative

    You can in fact pass along any CallerID you like and it will be then passed along to the end user if and only if the telecom provider supports it and chooses to pass it along based on your past history of using good numbers. Any telecom provider (VOIP or Traditional) can just reject the numbers and replace it with the main number of the business.

  7. Re:Does "open" include the ability to spoof caller by teasea · · Score: 1

    Alright, I don't claim enough knowledge of telecommunications to respond to this. It appears to be security through obscurity. I say we try an experiment and offer knowledgeable opinions as to why or why not we should treat tcom differently from other open systems.

    Please write clearly as I intend to plagiarize your statements for my thesus :)

  8. I've been watching too much Heroes lately... by Anonymous Coward · · Score: 0

    I read, "open source telepathy gives customers control".

    1. Re:I've been watching too much Heroes lately... by kryten_nl · · Score: 0, Offtopic

      When pull becomes push, there is no limit to what we can do.

      --
      For the perfect anti-Unix, write an OS that thinks it knows what you're doing better than you do and let it be wrong.
  9. Re:Does "open" include the ability to spoof caller by El+Pollo+Loco · · Score: 2, Insightful

    fear what phishers/advertisers/malware distributors might be able to do

    Well, the flipside is that of security. Open code will reveal bugs. These bugs will be fixed, and the code will be more secure. Don't forget that companies selling platforms on asterik have to support their products. Customers get mad when their telephones go down from hacking. The companies that sell asterik will have an incentive to fix the problem.

  10. Something funny... by Anonymous Coward · · Score: 0

    Linux.com doesn't display under IE7, so I tried it under FireFox. It loads, but there's a Windows Server ad on the front page... Wow

    1. Re:Something funny... by Bryansix · · Score: 0, Offtopic

      Linux.com doesn't display under IE7
      Yes it does. I just loaded it.
  11. It's more than hard enough already ... by Tim+Ward · · Score: 2, Interesting

    ... 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.)

    1. Re:It's more than hard enough already ... by domatic · · Score: 1

      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.

  12. Policy in action by Anonymous Coward · · Score: 0

    yes, this sytem was found to perform brilliantly here until disaster struck

  13. Re:Does "open" include the ability to spoof caller by Anonymous Coward · · Score: 0

    Caller ID is ancient obsolete tech anyway. The question for the future: can they forge RSA signatures? Probably not.

  14. Can be done from closed source PBX as well by transporter_ii · · Score: 3, Informative

    I might point out that you can change the number displayed by caller ID in many PBX systems, not just open source ones.

    transporter_ii

    --
    Doctors destroy health, lawyers destroy justice, universities destroy knowledge, religion destroys spirituality
  15. More on caller id spoofing from Slashdot by transporter_ii · · Score: 3, Informative


    Slashdot | Caller ID Spoofing Becomes Easy
    http://hardware.slashdot.org/article.pl?sid=06/03/02/2311218

    ID Spoofing for the masses:
    http://slashdot.org/article.pl?sid=04/10/28/1450205

    Slashdot | Caller ID Falsification Service
    http://slashdot.org/article.pl?sid=04/08/30/1620237

    New Google Service Manipulates Caller-ID For Free
    http://yro.slashdot.org/article.pl?sid=06/11/18/2112248&from=rss

    --
    Doctors destroy health, lawyers destroy justice, universities destroy knowledge, religion destroys spirituality
  16. IP Telephony Confuses Me by StaticEngine · · Score: 1

    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 search on the web, all I see are lots of paragraphs talking about "solutions" and no "here's what you need: one of these boxes, these cables, these phones, this service plan."

    Assuming I am not a LINUX DIY guy, and I don't want to hire someone specifically to manage telephony in addition to the normal business IT guy, where do I go for an off the shelf, or at least easily assembled, IP Telephony solution for my new company?

    1. Re:IP Telephony Confuses Me by MrMunkey · · Score: 1

      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

    2. Re:IP Telephony Confuses Me by Buelldozer · · Score: 1

      Call Cisco.

    3. Re:IP Telephony Confuses Me by lurcher · · Score: 1

      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/ it helped a lot, its biased towards asterisk, but also explains a lot of the terminology

  17. Re:Does "open" include the ability to spoof caller by dinofile · · Score: 2, Informative
    I understand your fears, but really this is ye old security by obscurity argument plus PBX's are not main exchanges.

    A home/smb PBX still has to connect to a main secure backbone exchange and as far as I remember they should validate your allocated range of callerID numbers or simply only allow a single switchboard callerID to be sent out. ISP's and Telecom's companies like cut deals and flog "authorized/validated/authenticate" hardware, but really that is more about signals strengths and kick backs than a secure box. User premises equipment must be assumed to be inherently hackable - end of story. The server/exchange side interface (ie. trusted telephone exchange) is where hacks are supposed to be caught.

    Sure in theory you could spoof the exact callerID within your range, but you can argue any 3rd party home PBX system could be capable of that. At least if the source is open, there is more change of reviewing whether it is definitively possible or not, before or after an incident. With closed systems, you are completely trusting the vendor and even with court action it would be hard if not impossible to get access to their source to prove it either way.

    What your home/smb does with its local PBX side is its own biz and you have to trust that yourself, just as you had to blindly trust some 3rd party PBX manufacturer of the past.

  18. oh, it's tele-PHONY by edjusted · · Score: 1

    I was skimming my rss reader and thought it said open-source tele-PATHY...darn...

  19. Not Everything... by mikecx · · Score: 5, Informative

    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.

    1. Re:Not Everything... by deniable · · Score: 1

      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.

    2. Re:Not Everything... by Galactic+Dominator · · Score: 0
      I found the SLA to be not so bad. It depends on the SIP clients you chose. In my case they are snom 360's, but which type of SLA are speaking of? The SLA done by assigning users to ZAP channels is very inefficient way of setting up a phone system.

      Or are you referring to SLA appearance offered by the asterisk clients? The different phones you speak of may have different UI's to initiate the SLA appearance, however at the SIP level the transfers happens with re-invites which is nearly universal and not in any way asterisk's problem(other than handling it correctly of course).
      http://www.ietf.org/rfc/rfc2976.txt

      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. In that long run on devoid of whitespace, I think I saw a complaint about asterisk not using enough ram. Or was it that it uses text based config files and not a gui?

      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. Welcome to opensource, however I'll point out that of the PBX's you pointed out, Asterisk is the only one you need worry about. All the others are just different kinds of makeup you can apply to asterisk, like changing faceplates on your 360. Since there are regular and substantive updates to asterisk, I'm struggling to see your point.
      --
      brandelf -t FreeBSD /brain
    3. Re:Not Everything... by mikecx · · Score: 2, Informative

      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 agree this is silly but in a smaller business this feature really is crucial. I know I could setup ring groups or have everyones phones ring and then transfer or a few other options but why change what works well and why retrain a staff (be it small) when they already know how to use something that is pretty much industry standard and supposed in almost every pay-for version out there. I thought the goal of open source was to provide a free implementation that was just as good that people could see the code and work on the code. Telling people that the code is too hard to write so they should find another way and retrain people just isn't the open source spirit. Call parking is only a solution if you have a secretary at the same desk during all office hours but thank you for the suggestion.

    4. Re:Not Everything... by deniable · · Score: 1

      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.

    5. Re:Not Everything... by NullProg · · Score: 1

      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 guilty to lack of well defined feature descriptions. I also state that without a dedicated documentation department, it would be hard for any company to provide them (I work for a small custom retail software shop and our documentation sucks).

      For any inspired entrepreneurs out there...
      Team up a freelance programmer with a freelance writer and start creating manuals (Print/PDF). The programmer tells the writer what each feature of the program does. Start with Asterisk, Apache, Gimp, Audacity, InkScape, Samba and Scribus. I'd pay $5 - $10 US for those.

      Food for thought,
      Enjoy,

      --
      It's just the normal noises in here.
    6. Re:Not Everything... by mikecx · · Score: 1

      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 same general business area). As a technical user/programmer if something isn't the least bit obvious to me or most people not involved in the programming process there is probably something wrong with it. That said, the Asterisk GUI and it's lead programmer are both quite a bit ahead of the curve but as just one guy with a little help progress is obviously not quick on such a large project. Perhaps just setting up webinars or hypotheticals would do wonders for the asterisk / digium folks?

    7. Re:Not Everything... by njhunter · · Score: 1

      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 Director without a sufficiently experienced Linux crew, so I'm taking the easier route and going with Fonality. I get full access to the Asterisk box at my site but all configuration is done with their GUI. And most importantly with their support. I had no idea how much a phone systems cost before I arrived at Asterisk. I had no idea that one could spend hundreds of thousands of dollars on a phone system. Granted companies are starting to look at the SMB but they are still locked away in the modules market. Take a Toshiba, for instance, you have a box that houses the cards, you want to upgrade, you get another card. You fill that cabinet and you have to buy a new card. You want to upgrade another part, well guess what, you throw away that cabinet and get a new one. Avaya has begun to think of the SMB and purchased another company. 3Com sounded somewhat OK but perhaps my rep got lost. Shoretel touted itself as way less than the competition but even they were in around a $100K. With all of the bells and whistles, Fonality was still half the cost. Straight Asterisk would be about a fifth of that. Fonality will give you the use of a demo system, it took me all of twenty minutes to call my rep with the demo handset using the new server. It sure helped sell management when I could show them that Intel was giving Fonality VC money. I have no connection with Fonality, other than being a happy customer. Knowing the underlying architecture, if I grow to hate them in the future, I still have an Asterisk boxes (thanks Markster!)

  20. Control is only to your door by sjbe · · Score: 4, Interesting

    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.

    1. Re:Control is only to your door by dumeinst · · Score: 2, Insightful

      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)

    2. Re:Control is only to your door by grasshoppa · · Score: 3, Informative

      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.

      --
      Mod me down with all of your hatred and your journey towards the dark side will be complete!
    3. Re:Control is only to your door by drspliff · · Score: 1

      (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 go for ISDN, E1 or similar.

  21. Some easy ways to get started with Asterisk by RCSInfo · · Score: 2, Informative

    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 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 machine.
    AsteriskNOW is made by Digium, the maintainers of Asterisk. Its still in beta, but there are prebuilt VMware and Xen virtual machines ready for download if you just want to give it a try.
    Elastix seems to be getting some good reviews, but the main site has been down for the past few days. The link to the left is the sourceforge pages.

    1. Re:Some easy ways to get started with Asterisk by Windowser · · Score: 1

      Don't forget http://centpbx.com/

      CentOS based with FreePBX and other goodies

      I dropped Trixbox when Fonality bought it and never looked back

      --
      Avoid the MS tax, always buy I.B.M. PC's (I Built-it Myself)
    2. Re:Some easy ways to get started with Asterisk by DataSpring · · Score: 1

      Another one that seems to be gaining a lot of traction and backed by a lot of online help, is PBX in a Flash - it is created and supported by Nerd Vittles.

      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 Vittles

      They also have some stuff you might not think of, such as setting up services on your Asterisk box to tell you the weather report for a zipcode you enter on your phone's keypad, or "MailCall" which allows you to get your e-mail via the phone, or a telephone reminder system... there are lots of options, and you can find out about most of them at Applications - The Best of Nerd Vittles

      There are also forums for lots of this stuff:
      Nerd Vittles Forum
      Voxilla VoIP Forum
      PBX in a Flash Script Site

      And, of course, the venerable VoIP Wiki at voip-info.org

  22. bingo by hurfy · · Score: 1

    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 ;)

    1. Re:bingo by wolrahnaes · · Score: 1

      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 what your POTS lines run you, but I've been replacing small PBXes and key systems with a hosted VoIP solution for 2.5 years now with no signs of business slowing. Either my company's sales guys are really good or it makes sense at some level.

      Certainly got to love the ease of troubleshooting a LAN versus analog stuff. Packet loss and jitter are easy to quantify and isolate. I run Linux-based routers to provide QoS and SIP optimization, so that has the side benefit of being able to tunnel the raw output of tcpdump over SSH in to a FIFO on my Mac which I then feed in to Wireshark, making remote diagnosis absolutely trivial.
      --
      I used to get high on life, but I developed a tolerance. Now I need something stronger.
    2. Re:bingo by knghtrider · · Score: 1

      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.

      --
      In America today you can murder land for private profit. You can leave the corpse for all to see, and nobody calls the c
  23. Re:Does "open" include the ability to spoof caller by Anonymous Coward · · Score: 0

    I fear what phishers/advertisers/malware distributors might be able to do it they are given too much access to the code

    Indeed, I reckon it's a damn good thing Microsoft keeps their source closed. The mere thought of what could have happened, if the source was open, makes me cringe.

  24. Re:Does "open" include the ability to spoof caller by maxume · · Score: 1

    No one who means it is relying on caller id for anything anyway.

    --
    Nerd rage is the funniest rage.
  25. Asterisk is FANTASTIC by dskoll · · Score: 1

    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)

  26. Why this stuff is not ready for Prime TIme... by FlyingGuy · · Score: 1

    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 is based in firmware and a lot of it is realized in hardware with custom ASICS and the like. Don't expect a general purpose computer to handle what they will. Yes these are all based on slip in cards but they also have to make some pretty severe trade-offs do to the architecture..
    • The software they run may very well be Linux based, but its not you average Linux. It might have the same kernel but that is where the similarity stops.
    • People trying to run their phone systems over things like cable modems are just nuts, you need a dedicated T1 channel for each line. Yes some people have had success with running things across data circuits that are cable based but there is nothing in the way of guaranteed bandwidth, its all best efforts read the contract. The biggest problem is that people want something for nothing these days, and complain when their voice is choppy when they are torrenting something or streaming video in either direction,
    • Modern commercial phone systems have millions and millions of R&D money that have been poured into them. The software has been QA'd up down left and right. The bug counts on the software are phenomenally low no matte who's system you are using, Mitel, Aveya, Panosonic, Northern Telecom, et all. Yes they are expensive, relatively, but when it comes down to it, measure that expense against not having your phones work for a long period of time because its OpenSource and bug resolution is slow in comming, or your el-cheepo ISP has some equipment failure.
    • The bottom line in telecom equipment is you get what you pay for.
    --
    Hey KID! Yeah you, get the fuck off my lawn!
    1. Re:Why this stuff is not ready for Prime TIme... by Anonymous Coward · · Score: 0

      Your an idiot. Try saving any config changes in IP Office Manager 6.0(10) without getting an error message. And before you get all bunched up, I know an upgrade would fix this "bug". But the fact that Avaya would release software that errors out 80% of the time on simple changes leads me to believe that R&D $$$$$$$ is being wasted on advertising...

    2. Re:Why this stuff is not ready for Prime TIme... by Anonymous Coward · · Score: 0

      he bug counts on the software are phenomenally low no matte who's system you are using, Mitel, Aveya, Panosonic, Northern Telecom, et all.

      I was with you up to the Aveya part. Their systems suck shit.

    3. Re:Why this stuff is not ready for Prime TIme... by dskoll · · Score: 1

      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 each line.

      Our 1-800 number comes in over our DSL connection and it works fine.

      Modern commercial phone systems have millions and millions of R&D money that have been poured into them.

      So does Microsoft Windows. Your point is... ?

      The bottom line in telecom equipment is you get what you pay for.

      No, the bottom line is that commercial PBX equipment is a great big ripoff and if commercial PBX vendors are not scared of Asterisk, they're in denial.

    4. Re:Why this stuff is not ready for Prime TIme... by FlyingGuy · · Score: 1

      Hmmm well I suppose I should forgive your insulting comments, but not so much today since I am in a bad mood, perhaps you were as well.

      Yeah I would agree that if the software they are running has problems then it should have never gotten passed QA and they deserve all the criticism for that.

      I will make the same statement I made in the end of my post, you get what you pay for. Phone switch manufacturers have been facing HUGE downward price pressure an no its not unique to them, but to stay in business they are forced to keep lowering the cost of the products and that is going to be felt at some level, and it appears that in Aveya's case it was felt at the QA level.

      I am old school and personally if I have a business that depends on phones I will purchase an appropriate bit of gear that has a solid reputation and provides years of solid service. I side as an exotic car mechanic and we have a sign in the shop that says:

      SPEED - QUALITY - PRICE
      Choose any two you like!

      --
      Hey KID! Yeah you, get the fuck off my lawn!
    5. Re:Why this stuff is not ready for Prime TIme... by Anonymous Coward · · Score: 0

      Your maxim doesn't hold water. Do you tell your boss to pick 2 when he asks you to perform a task?

      Your basically arguing that "free" software will never be as good as software you buy. That's something I will not agree with.

      As someone who is forced to support systems without service contracts (and a company and boss who won't authorize $300 for a 30 minute conversation with a support person) I constantly find myself in a position where I wish I had a test system (which is cost prohibitive if your buying the equipment) and/or an active and open community where I can learn more about the systems I support (and contribute to when possible). All equipment and software, regardless of its origins, will have bugs and issues at some point. Just because it cost more doesn't make it better or guarantee less problems. Most issues I've encountered, in regards to computers/phone systems/free software/everything I support, have to do with how the equipment/software is being used, not where it came from or how much it cost.

    6. Re:Why this stuff is not ready for Prime TIme... by FlyingGuy · · Score: 1

      Actually it does. Now before you scream how wrong I am lets take a few examples:

      • Gimp - Damn good but always loses when compaired to PhotoShop.
      • OpenOffice - damn good but always loses when compared to MS-Office or even Word Perfect office, this includes all of the parts, database, presentations, etc,. And just to be clear. I loath MS-Office and MicroSoft.
      • Firefox - Often compared to IE and comes up just short for whatever reason. I don't use IE except for testing as I much prefer Firefox.

      And yes I tell my boss ie: my customers, quite often you can only choose two. If you want software fast and cheep it is only going to do the bare minimum to accomplish the task, no bells, no whistles, and there will be very little abstraction as that takes time, thought and planning. It is going to be hard coded for the most part and in the end you will end up paying more for it. If you want it yesterday and you don't mind paying a premium I will bring on extra crew and it will get cranked out fast with bells and whistles. If you want quality and low price then you are going to wait and I will fit it in when I can and not charge you as much.

      --
      Hey KID! Yeah you, get the fuck off my lawn!
    7. Re:Why this stuff is not ready for Prime TIme... by Anonymous Coward · · Score: 0

      Loses...? You use FireFox but consider IE to be a better product? Better because you paid for it, not because it works better, and FireFox isn't as good although it has the features you find useful in your browsing?

      As a lowly employee I don't get to dictate the kind of relationship I have with my "customers". If I told my boss he could pick 2 of 3 he would can my ass. Don't get me wrong, that probably wouldn't be a bad thing.

      No offense, but your attitude reminds me all the old farts around here who feel like they've put in their time and know how things should work and refuse to explore alternative ways of doing things. I'm sure no matter where I go this will be the way things are. I guess that's why so many people start their own businesses/projects and try to do things in a different way, a way they find to be better.

  27. Tips by asphaltjesus · · Score: 1

    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.

    --
    Got Trader Joe's? friendwich.com RSS feeds work now!
  28. See your ISP by asphaltjesus · · Score: 1

    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.

    --
    Got Trader Joe's? friendwich.com RSS feeds work now!
  29. Re:Frost dring dring by The+Clockwork+Troll · · Score: 1

    Asterisk? The gall!

    --

    There are no karma whores, only moderation johns
  30. Not Everything...If Safeauto did manuals? by Anonymous Coward · · Score: 0

    "As a programmer, I plead guilty to lack of well defined feature descriptions. "

    As a writer I plead to...Oh wait. We're talking about your shortcomings weren't we?

    "Team up a freelance programmer with a freelance writer and start creating manuals (Print/PDF). "

    Somehow I'm reminded of that old saying, "Linux is free if your time is worth nothing". Wonder if that applies to free documentation? And yes you said $5-10, but I think you'll find quality documentation that would cover the (small) demographic Asterisk is aiming for would cost a bit more than that. Same with the others on the list. And lest you forget it takes more than a writer (even several) to produce good documentation.

  31. Take it form a Pro by Anonymous Coward · · Score: 0

    Its (VoIP)not ready for prime time.

          Imagine your business being dependent on independent data corridors which carry telco traffic globaly, not unlike our system of highways and in many ways analogous to these same highways controlled by individual states. Degregulation was great but not without cons. The old style monopolies had one advantage over the fragmented hodge podge of intra lata carriers we have today, centralized control of all high bandwidth trunking and localized or last mile loops.

          What this means to you the customer is really simple, decentralized management by independent entities all vying for their own independent customer base have a tendency to oversubscribe their services resulting in a lack of adequate real time resources ala bandwidth. They continually oversell access and its the VoIP customer who is last in line due to the nature of packet switching and the existence of the traditional circuit switched services.

          Yes these data highways are all interconnected but one experience is universal and a distributed phenomenom, bottlenecks aka traffic jams that are not centrally managed. What does this mean to you, simply, unintelligble phone calls from customers or customers who cant get through.

        Bottom line is that your subjecting your business and its future to this jungle that is telecom. I have worked in the industry for almost 20 years and in that time have seen customer disatisfaction grow and mostly with VoIP.

        The funny thing is that people have a high tolerance for a crappy sounding phone call until their level of activity reaches a point where they are spending too much time switcing carriers or logging complaints while pissing off their customer base.

          With all of that time and aggravation, any savings realized from VoIP was lost in addition to the cost for blood pressure meds.

          So bottom line, Circuit Switching, the premium grade telecom voice service, is here to stay and it will take a quantum leap in processing to increase bandwidth a significant factor over what it is today to allow VoIP to truly compete quality wise.

          Until then, enjoy your jitter, echo, distortion and lost connectivity and dont call me, I am too busy supporting this frankenstein.