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Solutions for Small Business VoIP?

Posted by Cliff on Tue Nov 29, 2005 09:18 PM
from the from-copper-to-network dept.
MajorBlunder asks: "I'm part of the IT department of a small but prospering software company. We have recently filled the capacity of the POTS PBX phone system we currently have installed. We are currently looking into switching over to a VoIP phone system. We have a sizable IT staff in proportion to the rest of the company, so we'd like to be able to maintain the hardware/software in house as much as possible. I wanted to ask the Slashdot readership what experiences they have had with switching over to from POTS to VoIP. Any recomendations for full end to end solutions would be appreciated, and recomendations of things to avoid would be great."
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  • My experience (Score:5, Informative)

    by 2.7182 (819680) on Tuesday November 29 2005, @09:21PM (#14143685)
    I have a small printing shop that switched 6 months ago. Our first thing was to make sure your bandwidth settings were set to the highest value. This can be set on the Vonage website and I last I looked there were 3 choices. I have seen new lines default to the lowest setting which is total crap. I have 3 lines on a cable modem connection and have never had call quality issues. I have had just about every other issue with ringing and connect delays, voicemail, caller id, etc. Most of the time you pick up and say Hello and the other person doesnt hear anything cause the call has not properly connected yet. But it saves me hundreds/month and the minor issues I have learned to live with. --
    • Re:My experience (Score:5, Informative)

      by XorNand (517466) * on Tuesday November 29 2005, @09:52PM (#14143841)
      Setting the call quality to the highest setting means that the G.711 codec is used, which consumes 64k/s per conversation. That's generally not a problem with a home user who only has one call happening at a time, but it will easily overwelm the standard small-business broadband connection which might only have 128-256kps upstream bandwidth. Setting the call quality lower is probably using the iLBC or the GSM codec. GSM is commonly used for cell conversations, iLBC is a variable rate codec designed for VoIP. They both consume far less bandwidth, but you're right, the call quality sucks.

      An alternative is to use the G.729a codec, which is almost as good as G.711, but only uses 8kbps per channel (plus TCP overhead). This is a far better solution, but the reason you don't seen VoIP providers offering G.729a is because it's patent protected and therefore requires that the provider purchase a license for each concurrent channel in use.

      Ugh... I really wish this topic got posted next week isntead of now. Forgive the blatent plug, but I've recently started a VoIP service that caters exclusively to small-businesses and solves the exact problem presented in this thread. It's similar to a Vonage-type setup but we support G.729a, plus all the features of a business phone system (voicemail, auto-attentant, transfers between extensions, etc). All of the systems engineering is done and tested and we're accepting customers, but our website won't be unnveiled for another couple of weeks. Five extension plans start at $224/mo. and scale up to 25 extension plans. We're focusing mainly on offering the plans through a network of small VAR resellers who want to earn a monthly commission. If anyone wants more info, drop me a line at resellers@brightideavoip.com.
      • Actually, Bandwidth In Mirror Will Be Larger Than It Appears (BIMWBLTIA)! And, when it gets right down to it, you don't care about bandwidth anyway; you only think you do.
        1. Why do companies spend $500 a month for a 1.544Mbps T-1 when a 1.5Mbps DSL connection is only $29? BECAUSE YOU DON'T CARE ABOUT BANDWIDTH (you only think you do. more below.)
        2. Why does your 64Kbps codec consume more than that when you actually look at it? BECAUSE OVERHEAD COULD DRIVE THROUGHPUT AS HIGH AS 3,500Kbps! (actually that's ju
    • I predict 37 clueless "Vonage" replies before this thread reaches 100 comments. 34 of those users will continue to defend their clueless "Vonage" answers even after it's pointed out that they don't want to ditch the landlines so much as they want to have more extensions than their phone hardware allows. Of those 34, 29 of them will have no clue what an extension is in this context, even though they have certainly dialed an extension at least once in their useless gibbering idiot lives.

      BTW, for those of you
    • Re:My experience (Score:4, Interesting)

      by neilticktin (660748) on Wednesday November 30 2005, @12:31AM (#14144675) Journal
      We're getting ready to do a cover story in our magazine about our experiences with VoIP. To do this, we decided to "eat our own dog food" and move the entire company to VoIP.

      In short, I'm glad we're on VoIP. We're using a smaller provider, which gives more personalized service ... and that's been a big win. The company is PhonePipe ... www.phonepipe.com ... and aside from the usual bumps in the road, we've been glad that we went with them.

      A few things to consider. Some VoIP companies are not financially stable, and they many times don't fall under the FCC rules. So, you should check out the companies you are dealing with ... even some of the biggest ones are not financially sound.

      For hardware, go with either ATAs or the Cisco phones. ATAs will allow you to preserve your prior investment.

      Lastly, be aware that you may need to do some traffic shaping, QoS, etc... And, that many times, the cheap consumer routers handle VoIP much better than the higher end stuff (believe it or not).

      Favorite features? Simultaneous ring, and the ability to filter which calls get through and which get routed right to voicemail.

      Good luck with it!

      Thanks,
      Neil Ticktin
      Publisher, MacTech Magazine
  • The obvious choice. (Score:5, Informative)

    by killjoe (766577) on Tuesday November 29 2005, @09:21PM (#14143687)
    Go to the digium web site, pay them a thousand dollars, and let them install asterisk for you. Either that look around for a local asterisk provider. If you live in a metropolitan area you should be able to find a few without any problems.
    • by kasparov (105041) * on Tuesday November 29 2005, @10:25PM (#14144019)
      Although it would be nice to give Digium some money, for a company that has a good sized IT department it is unnecessary. Asterisk isn't particularly difficult to get running. Going through the setup and configuration could come in handy if they are planning on maintaining it as well. And, if they are really lazy, they can use the Asterisk Management Portal [coalescentsystems.ca] or even Asterisk@Home [sourceforge.net] (which uses AMP, but includes some other features).

      The poster didn't mention how many phones/lines they need, but if they need to they can use VoIP internally (for unlimited internal phones), and just hook up T1s from the POTS for as many voice lines as they need (if they are worried about the voice quality/potential unreliability of VoIP providers). Digium has Quad-span T1 cards [digium.com] with onboard echo cancellation, so it should scale to the number of lines that are needed.

  • VoIP is not cheaper (Score:3, Interesting)

    by Py to the Wiz (905662) on Tuesday November 29 2005, @09:22PM (#14143698)
    ... at least for us (a small business). Once you add in all of the per-line charges, the hardware, the setup fees, the broadband, and the fact that if you want to use DSL, you still have to buy at least one phone line from the phone company. Plus, of course, the reliability of broadband still isn't nearly at the level of hard telephone lines. After taking this into consideration, unfortunately, going through the local Ma Bell monopoly was still the cheapest and most reliable option for us (a business needing 3-5 phone lines).
    • by Trejkaz (615352)
      The calls themselves are most certainly cheaper, though, so I suppose it really depends whether you make a lot of calls, or hardly any calls. If you consistently make interstate calls then there would be a big difference between paying STD rates for every call, vs. paying a tiny flat rate for every call.
      • Many small businesses don't have a T1. In many areas, DSL/Cable modems are not even close to reliable as a T1 from a prominent provider. Also, the cost differnce between a T1 and DSL/Cable line is usually quite significant, and most often the DSL/Cable connection will provide much better bandwidth.

        In my case, the (2mbps/768kbps) DSL we had was horribly unreliable. We switched to Cable and while it's been reliable enough to use it for VoIP, to buy the voice lines from the Cable company isn'
  • similiar position (Score:5, Informative)

    by sgeye (757198) on Tuesday November 29 2005, @09:29PM (#14143729)
    I work for a small firm, 100 people or so across 3 offices which are relatively close, about to add another 20-40 people. We are in a similiar position, because our old PBX system won't handle that many users without some upgrades, which we don't want to do because it is reaching the end of its lifecycle. We did a little looking around, and suprisingly the Cisco Call Manager Express was the best priced solution for us. The only way we could beat their price was going with an IP PBX system instead of a VOIP solution. They were running a promo, so there was a 39% discount from the list price on all hardware. Unfortunately, the owners decided to hold off on the upgrade and bandaid our system until late next year because we will be moving into a new building and merging two of the offices. We couldn't get a quote from Avaya, their rep never called us back, and both 3com dealers we spoke with had recently quit selling 3com. I can tell you not to go with Nortel, their solution was over 1.5x that of the Cisco solution.
  • by PogiTalonX (449644) on Tuesday November 29 2005, @09:30PM (#14143734)
    I work for a company that has about 12 people and we use the Cisco Systems [cisco.com] IP Phones. They work pretty well, have all the features of a normal PBX including intercom, call transferring, etc and they're relatively cheap.

    The cool thing about these phones is each phone gets its own real phone number as well as internal extension. We are located in California and when we have trade shows in Florida we take one of these phones and plug it into any ethernet jack. The phone auto-configures itself and you get the same phone number and extension and you can call other people in the office on speaker as if you were in the next cubicle. Pretty rad. Hope this helps.

    • Cool, but I see two issues here :

      1. If you just plug your phone in any Ethernet port and get connected, that mean your VoIP is accessible at large. Personnally, I would not make my PBX reachable from the Internet.

      2. Hopefully, your phone use some kind of encryption for the signalling and voice transmission. Not all do, don't know about Cisco.
    • The cisco phones are nice, but the feature you reference is actually called DID (Direct Inward Dial) and is available with almost any digital phone service (CAS/PRI and of course VoIP). Basically it lets the office have a bunch of numbers that will ring into the office's PBX main number, and lets the PBX decide where to route them based on a certain number of digits sent from the actual number dialed, which is why your extension is probably the last few digits of your desk's full number. When you dial out,
  • Put them on their own network segment. Also, if you'll use them in a mission-critical capacity (like a call center), make sure you keep in mind that if the network goes down, so do the phones.

    Lastly, your price per phone is going to be somewhat higher.

  • by lkcl (517947) on Tuesday November 29 2005, @09:32PM (#14143747) Homepage
    okay, here's where lots of VoIP things go wrong: they think it's okay
    to use the same line for normal internet access as well as VoIP (i'm
    assuming you have a broadband line with an upload speed of max 256k
    but this also even applies - if you load it enough - if you have e.g.
    1MB SDSL).

    given that the MTU has to be slammed up so far (in order for ISPs to
    compete on "bandwidth" rather than "latency") to ridiculous levels
    (1400-1500) it leaves very little options at _your_ end even if
    you _do_ do QoS tricks.

    so, your only _sensible_ option is: get a second broadband line,
    and use it _exclusively_ for VoIP.

    and if you are going to do _that_ then make sure that you get a fixed
    IP address and put the damn ADSL card _in_ the asterisk [or SIP] server.

    the reason is quite simple: NAT on SIP is a _complete_ bitch to set up,
    especially due to RTP (the audio) and you can avoid an awful lot of hassle by putting the ADSL card
    into your server, so it is a direct interface on the server. this assumes,
    of course, that you're not running windows!

    also - make sure you use 8k CODECs like GSM, because you very quickly run out of bandwidth
    on a 256k upload if you use 32k CODECs.
    • Don't use GSM, use G.729. I recently switched from softphone/G.711 to PAP2/G.729 and the call quality is much better. I was getting complaints of sounding like I was in a tunnel or on a mobile, but people can't tell any difference with this new setup.

      And if your VSP supports IAX then there will be far less overhead. (Can then run X number of calls with 1 set of overhead, instead of X number with X sets of overhead with separate SIP lines).
    • you can do QoS - and ask it to prioritise SIP and RTP packets. however, RTP is a pain: the _clients_ decide which damn range of ports they will go out on, so you need to use a sip proxy to "rewrite" the SIP/RTP packets to be within a certain range (apt-cache search sip proxy if using debian - don't bother with anything else).

      so, you've installed a sip proxy, it rewrites the RTP packets so they only go out on ... say... ports 10000 to 11000, and you can set your QoS to prioritise any UDP traffic on those po
  • One of the best things you can do is get managed switches. If you have remote users, don't cheap out on the VPN endpoints. Expect some "echo".

    I work on the data side, not the phone side of the company. If we had "paid" for our system, I'd be pissed.

    I'm not familiar w/ Asterisk which has been mentioned. We only deal in a commercial offering, by a *huge* electronics company. Our main phone tech says, "you *are* going to have some problems w/ VOIP over the internet. As long as you keep it in-house, w/ th
  • asterisk (Score:3, Informative)

    by max born (739948) on Tuesday November 29 2005, @09:35PM (#14143762)
    Try asterisk [asterisk.org].

    Just playing around I set up a 10 extension inter office VoIP system using this system in about 20 minutes on an old laptop. It's open source, free, and has a great a community behind it.
    • Just playing around I set up a 10 extension inter office VoIP system using this system in about 20 minutes on an old laptop. It's open source, free, and has a great a community behind it.

      Hey, I'm sure you really did achieve this in 20 minutes (OK, I'm not sure at all, unless you'd already done it a few time before...but who knows...)

      But I'll just add a voice of reason here that asterisk, while it definitely is a great solution and has a fantastic community, is a real sophisticated system that may well

  • BYOD @ Broadvoice (Score:5, Informative)

    by TheRealFritz (931415) on Tuesday November 29 2005, @09:40PM (#14143782) Homepage
    I've switched to using http://asterisk.org/ [asterisk.org] along with http://www.broadvoice.com/rates_compare.html [broadvoice.com]. I think you'll find this Wiki to be a very useful resource: http://voip-info.org/ [voip-info.org]

    The plan I'm using is BYOD-Lite which costs me only $6 a month and there was no activation fee, since I had my own VOIP equipment in the form of an Asterisk PBX installed on Linux. From what I can tell, they are one of the few providers who allow the use of customer supplied VoIP hardware/software, in my case Asterisk.

    Something you'll have to research is what technology you want to use for hooking up individual phones to Asterisk. One possibility would be to use hardware from Digium: http://www.digium.com/index.php?menu=product_categ ory&category=hardware [digium.com] or any other Analog Telephone Adapter (ATA), or you could use Softphones installed on employee PCs such as X-Lite (free), or similar.

    Good Luck!

    http://www.gloryhoundz.com/ [gloryhoundz.com]
  • VoIP can be tricky - stay away from going exclusively VoIP, for example using Vonage, Broadvoice etc... for business in my experience it's just not there yet. The trickiest part will most likely be choosing the right phones and integrating with whichever PBX / Gateway you'll be implementing. Asterisk is a very solid option - but make sure the server that it's running on is reliable and that the IRQ issues aren't a concern with the hardware.

    Getting outbound VOIP Lines might not be mature enough for your c
  • Unless you know enough about VOIP to setup your own. Remember, you're going to be maintaining this over and above your current job functions. It may or may not be benifical to go with something like Asterisk and going it alone. But, if you do go with a consultant, for the love of God do NOT go with SBC. They setup our Cisco VoIP system, and screwed us by not giving us the discs and key codes to the CallManager or Unity software. They did leave the IPCC software in a corner cube, though.
  • by mflorell (546944) on Tuesday November 29 2005, @09:44PM (#14143805)
    last three years. We now have over 250 phones installed at 4 locations(including a call center). We started switching to Asterisk three years ago and grew the system to the point where everythign is Asterisk and we do all inter-office calls over VOIP(IAX trunks). The cost savings in licensing costs alone more than justifies 2 full-time IT staffers salaries.

    If you have some time to get comfortable with it, you will be very happy with the control you have over the system and the tremendous choice in phone hardware you can use with Asterisk. And if your company is anything like ours, they will love the cost savings.

    Here's a link to a case study presentation I gave at Astricon 2005 last month:
    http://astguiclient.sourceforge.net/astricon_2005/ Florell_astricon_2005.html [sourceforge.net]
  • by g-san (93038) on Tuesday November 29 2005, @09:44PM (#14143808)
    Your network is a factor here as well. Do you know how much traffic you have on the network currently? Can your routers do prioritization on different traffic types, either IP Type of Service or tcp/udp port? You want to have that understood to make sure the quality is good, so VoIP doesn't affect your usual traffic and vice versa.

    You can also get switches/modules nowadays that have Power over Ethernet (PoE). So of the two RJ-45 connections (you have the physical cabling for this, right?) in a cube, one connects their PC and the other connects the VoIP appliance/phone back to the PoE port. The phone gets it's power from the ethernet cable. If those switches and the rest of your key servers and network are on UPS, the phones still work when the power goes out.

    Good luck.
  • Did you not see... (Score:3, Informative)

    by syukton (256348) on Tuesday November 29 2005, @10:03PM (#14143894)
    Did you not see this story the other day [slashdot.org] about the new open source magazine, O3 [o3magazine.com]?

    Their first issue [o3magazine.com] "looks at reducing voice infrastructure costs with open source telephony solutions"

    I suggest starting there.

  • Shoretel (Score:2, Informative)

    http://www.shoretel.com/ [shoretel.com] - makes the best VOIP phone system around. It will do everything you want it to do, easy to set up, and comes with an SDK. Knock yourself out.
  • RHEL3-ES ($349) + Asterisk Business Edition ($995) + Dell PE2850 (~$5000, dual 2.8, 1gb, raid1-76gb, 3yrs-4hour-onsite, drac, redundant-psu) = $6,500ish

    Thats not counting phones, network upgrades, and whatever cards you'll need for your asterisk box to talk to things. So figure 10K.

      • Jesus that's a big server for Asterisk. I've pinned up 600 calls / 60 cps with RTP (mind you, ulaw) against the echo app and sat at an average 70-80% idle on a modest old dual Xeon.

        Codec and transcoding is everything when it comes to Asterisk and CPU. Try running the same setup with g.729. Hint: My box with dual 3.6 Xeons max out at around 120 calls when it needs to transcode g.729 for pstn termination. If Asterisk only needs to pass the packets along without transcoding then it can handle thousands of ca

  • by RedLeg (22564) on Tuesday November 29 2005, @10:33PM (#14144062) Journal
    Asterisk is more than likely the ultimate solution to your problem.

      - The bad news is that it has a VERY steep learning curve, that is unless you are expert in linux, telephony, and a few other odd disciplines, a relatively rare combination these days.

      + The good news is that you can test drive and get up and running quickly and cheaply with Asterisk @ Home..

    Google for Asterisk @ Home. D/L the CD, take a SPARE box, one that you have no residual data on ('cause it's going to get zorched), insert the CD and follow the prompts. About an hour later, you will have an installed and (mostly) configured PBX with a web management GUI and a huge support community.

    Believe it or not, you can install it in VMware and get a good feel for the functionality without sacrificing a box or boxen to the PBX gods.

    The project is extraordinally well documented, and the only additional things you absolutely need to get started playing around are a soft phone (or an IP phone, or a ATA and an analog phone) and a Freeworld Diallup (no charge) account. A cheapass PCI card to connect to a single POTS line will run around $10 on E-pay.

    All of this will take no more than a couple of hours, and you should be able to get a really good idea of what Asterisk is capable of doing.

    Once you've convinced yourself (and your colleagues), you have some choices, namely, build it yourself or buy. I can't offer advice here.....

    Some other potentially useful info-tidbits:

    • IP Phones are readily available starting at around $45US a set for cheapies (new, but low frills and crappy docco), up to several hundred a set for top-o-the-line units from folks like Cisco. I would personally recommend at least two or three for your pilot project, and not all the same model.

    • Beware the "power adaptor problem.' Some VoIP phones are designed to use POE (Power over Ethernet), where the switch provides the power over the ethernet cabling just like the phone company. If the phone sets are designed for this, they may not come with power bricks, and these particular bricks can be very expensive, and add considerably to the cost of the phone set.

    • ATAs (analog telephone adaptors) let you plug a phone (or a fax, or both) into an ethernet link connected to a VoIP lashup. These are what a LOT of the commercial VoIP providers furnish or provide at low cost. There are LOTS of these available on the secondary market, and many can be unlocked to use with any provider. I'd recommend you play with a couple different ones of these as well.

    • There is a metric a$$load of information on VoIP, Asterisk and Asterisk @ Home at VoIP-Info.org [voip-info.org]. Among other things, you can find info on which phones (soft, hard and ATAs) are well supported, and config info for lots of specific models.



    Hope this helps.....

    --Red
  • by mlg9000 (515199) on Tuesday November 29 2005, @10:37PM (#14144081)
    VOIP is a buzz word right now but it usually doesn't make sense. A T1 will carry 16 VOIP calls (at ~POTS quality) and runs ~$400 a month. A PSTN line (T1 for voice) carrys 24 lines and costs ~$350. VOIP phones cost almost twice as much as digital POTS phones. Plus there will be a cost going from POTS Minutes are slightly more expensive with POTS but you'd have a use a whole hell of a lot of minutes before you'd hit the break even point. So unless you are a heavy user it doesn't make sense. If you had multiple locations and needed internal extensions etc that might work too. Site to site data lines are much cheaper.
    • by anticypher (48312) <anticypher @ g mail.com> on Wednesday November 30 2005, @05:39AM (#14145678) Homepage
      A T1 is 1.5 Mbps. Using a reasonable quality codec like G.729ab means you can fit 85 to 100 simultaneous calls into a single T1. Certainly you could stick to G.711 a/u-Law codec and have slightly better quality than G.729ab, and even with signalling overhead (either H.323 or SIP), you could fit 22 simultaneous calls into a T1.

      These numbers comes from a real, working system. It's right now passing 85 calls, and consuming 1.5 Mbps. This particular VoIP router is sitting on an E1 (2Mbps) and can pass a maximum of 120 calls.

      Are T1 circuits in the U.S. still so expensive? Do carriers charge more for an unframed data circuit than a PRI phone circuit? (which sounds bassackwards, but it's the new unregulated America where anything can happen) Average price for an E1 in Europe is about US$150/month for a data circuit, and depending on the phone company at the other end, about US$250/month for PRI over E1.

      the AC
  • Managed VoIP PBX (Score:3, Informative)

    by segment (695309) <sil.politrix@org> on Tuesday November 29 2005, @10:40PM (#14144095) Homepage Journal
    Currently we are using Covad after a horrendous experience with Packet8 whose Virtual Office product line is nothing worse than your worse thought. I have 8 offices spread through the US and wondered about setting up Asterisk even went as far as having them quote out a prebuilt drop in system. The problem with this became the cumbersome syntaxing of Asterisk. I don't mind, nor does my coworker but it is not a feasible system unless you have experienced engineers in those offices when a problem arises. Sure you could talk about KVMOIP to manage issues but sooner or later you will need someone to touch that machine. Anyhow, experiences with Asterisk: echo, cancellation issues and all that fun stuff. For example if you're using a Digium card you will need to up it to about 256 taps. A tap represents 1 sample, and @ 8kHz (which is what all of Asterisk's echo cancellers default to) each tap represents 0.125ms. Asterisk default of 128taps will therefore handle echo paths of up to 16ms, supposedly good for most things. You may get better results with fewer taps cause training time is shorter and the canceller will adapt faster. Conversely, if you're having problems with echo on long-distance phone calls, you may need to up this to 256 taps. BUT... Asterisk only lets you set 32, 64, 128 or 256 taps. Using a different number of taps will cause Asterisk to revert to 128 taps without warning. So if you can't get echo out @ 256 you're going to have a handful of daily complaints on echo using Asterisk... Outside of that funkily chopped and pasted information, physical phones. What kind of switches, your speed, and all other even funner (is that a word funner) things come into play. Will you have an allocated connection for phones? Sure you would not want to have the lines on the same lines as your Internet data lines. Think of the costs behind that. Phones physically, I'm not impressed with too many VoIP phones. Right now I have Cisco 7960's and 7940's, and those supposedly are top of the line which still don't impress me much.
  • by donnacha (161610) on Tuesday November 29 2005, @11:10PM (#14144211) Homepage
    Think before you leap because the potential of VOIP is tantalizing, believe me I know, I got sucked in and, to be honest, in many ways I regret it.

    I'm a home user/home worker, none of my calls are that important but the quality definitely isn't there. We humans have a great capacity to blind ourselves to minor inconveniences, such as having to alter our conversational style to accommodate slightly unsychronised conversations or drops of several seconds in which the other person can't hear us but, ultimately, these things wear you down and change your relationship with your phone - you can no longer trust your phone but, like the flaws in a new lover, you excuse these things because you're so enamoured with the promise, the potential to route around the bastarding telephone monopolies that have held us all hostage for so long.

    I should mention that I'm a UK user and, obviously, that places an extra burden on a US-based service. I signed up to Broadvoice because they had the best thought out plans and their support is, well, it exists which is more than can be said for many of the others. On the whole, though, I absolutely cannot recommend them to UK users because they let me down badly with regard to 0800 (UK tollfree) and 0870 (UK region-free numbers) which, although they claim otherwise on their rates pages, they simply cannot connect to, not for any amount to money. This alone renders their service redundant because, in the UK, an increasing number of businesses only provide and 0800 and 0870 number. The best example of this is Apple's UK branch who no longer accept emails - I wanted to buy about £3000 worth of computers and emailed them with a query, received an automated reply telling me that the only way to contact them was via their 0800, with no regular number to use as an alternative. This may sound like a fairly marginal problem but you wouldn't believe the number of times I've ended up using a mobile, at 20p per minute, to wait on a "freephone" service queue. Apple, BTW, lost that sale along with the chance that I'll ever again suggest their systems to a client.

    So, for home users looking to save a few quid, don't buy into the dream while it's still a dream; certainly don't replace your main phoneline.

    For home workers attracted to the idea of contacting clients all over the World, ask yourself if you, as a client, would be happy dealing with a service provider who you can't hear properly or with whom conversations are arduous.

    For executives eager to boost their corporate careers by manfully slashing millions from their company's telecoms bill, ask yourself if adding an extra stress to the every single employee who uses the phone might not be, in the long-term, a serious blow to the company as a whole - somehow added employee stress and customer frustration never makes it onto Powerpoint presentations, but it's smart to know what's annoying the Hell out of your rank and file.

    I wanted VOIP to live up to the dream, I really did - all I'm saying is that, in my case, it didn't, be aware of that amidst all the hype.
  • by coryhamma (842129) on Wednesday November 30 2005, @12:32AM (#14144684)
    One aspect of a VOIP system you may want to consider is the potential for redundancy.

    If you should happen to choose to go the Asterisk (open source) route, the Asterisk@Home distribution installs straight off a CD and can be backed up / restored through a web browser. This means that if you exclusively use IP connected components -- T1 or POTS gateways and IP connected phones -- then you only need to shove the Asterisk@Home install CD into another server should one fail and restore a recent backup -- voice mail, configuration and all.

    In addition, you can get a much higher level of service (potentially) from a service contract with an Asterisk consulting firm than your traditional Nortel / Toshiba / Avaya vendors. For example, if your phone system itself should suffer a meltdown, it is easy (in a small to medium office) to swap it with a PC. If a switch or T1 gateway should bite the dust, they are generally inexpensive enough to keep a spare around. My experience with the "big heavy" vendors is that a service contract will get you up & running in a day or less -- while a asterisk solution could potentially recover from the same type of hardware failure within an hour.

    I have to recommend against using a VOIP phone service however -- getting a T1 line from a good provider is likely to be cheaper and much more reliable.
    • Re: Asterisk (Score:2, Insightful)

      Having seen smart people struggle to get Asterisk working (cool a system as it is!), I imagine there would be quite a brisk market for a pre-configured, low-power box running asterisk ready for the user to plug in some custom messages, and / or rely on existing generic ones. That is, something truly plug-and-play, providing your have at least one POTS line to which it can be connected.

      Such a system needn't be *cheap* exactly in order to be quite a bit less expensive than typical PBXes, which are usually ove
    • Re:Asterisk (Score:5, Informative)

      by amliebsch (724858) on Tuesday November 29 2005, @09:49PM (#14143825) Journal
      Asterisk is definitely the definitive VoIP PBX-in-software, is FOSS, and runs on Linux. I've been testing it for a bit now, and it is a very nice, configurable, and reliable piece of software. If you use SIP phones, no additional hardware is required - the phones plug right into your LAN.

      Where it starts getting tricky is how to connect your LAN-phones to the outside world. You can use POTS lines, or a BRI or PRI, or a T1, but that all requires additional hardware from Digium. You can get VOIP service from many cable companies and CallVantage and Vonage and such but beware! If the VoIP service requires you to use their hardware adapters, you STILL need additional hardware. You might save a little money, but other than that there is no advantage for POTS if you have to use their adapters. Plus, what a kludge that is. Your incoming call goes digial(in)--> analog(adapter)--> digital(PBX)--> analog(phone)--> digital(PBX)--> analog(adapter)--> digital(out) JUST in your PBX! If you can get/can afford the bandwidth, a 100% digital solution requires minimal hardware investment (only the phones and the PBX server). There still don't seem to be that many providers, though. But I have had pretty good luck with a couple. Broadvoice [slashdot.org] has a BYOD (bring your own device) line of rate plans that are compatible with Asterisk, though you can only have 2 simultaneous lines per account. Teliax [teliax.com] has a flat-rate plan with up to 4 simultaneous calls, and you can have an unlimited number of simultaneous calls (subject to bandwidth constraints) using the Pay-As-You-Go plan. The other nice thing about Teliax is that it supports audio codecs other than the standard 64kbps(per incoming and outgoing channel) that Broadvoice supports. Using more efficient codecs will allow you to pack more simultaneous calls in the same amount of bandwidth.

      Oh, and use a high-quality router that supports QTos packet prioritization.

    • Re:Asterisk (Score:3, Interesting)

      by e4g4 (533831)
      I set up a small voip system in our office in NJ (3 lines) using broadvoice paired with asterisk - and while the service (most notably broadvoice tech support) leaves some things to be desired - our phone system is much better in terms of its feature set than it was on our POTS pbx. That said, most of the reliability issues we've encountered were the fault of our service provider, and we're generally quite happy with the switch.

      The website i found myself constantly referring to in terms of making phone,
    • by Anonymous Coward on Tuesday November 29 2005, @11:49PM (#14144431)
      What is the deal? All you have to do is link asterisk.org and you get modded up 4 informative? geeze, is that sarcasm in the mods???

      OK REAL Voip in a nutshell. You can run voip INTRAoffice then go out to copper (PRI) yourself or you can find someone to do voip trunking. (ie Your voice travels to an offsite virtual PBX and they send it to the pstn) [I say REAL voip because I'm talking business class, not running skype over a dsl line for kids to talk.]

      While trunking is the coolest way to do it, sadly, voip trunking is about where cell phones were in the late 80's. Useable but you had to be sorta dedicated to the task. But I'll give you an example.

      One of my clients decided to let speakeasy do the trunking. I (then) wholeheartedly recommended Speakeasy. It was a nightmare.

      The problem was that we were like their third business VOIP customer. The bigger problem was that they lied to us and told us they knew what they were doing. I've been a full time geek almost 20 years. --I have NEVER had a customer support nightmare as bad as speakeasy VOIP.-- The problem was they had nobody trained on the system and they just made shit up. Then when you asked them to do what they said they could do, they would claim they never said it. I got to the point where I put EVERYTHING in writing.

      If they had just come clean and said "Hey, we're learning this, give us a break" I would have helped them... But they didn't. I finally left my "dedicated" support person and went into the regular support queue. I got the support person to admit they were so new at it and they were clueless. I went back to my "dedicated" support person and told him the gig was up and he just stammered.

      ****But the service was good*****

      The fact they were lying sacks of shit not withstanding, by the time they delivered the product, it worked well.

      The topology goes like this.

      You have a Edgemarc router (I think it is edgewaternetworks.com, google is your friend) and you put everyone behind it. (Voip phones, workstations and even servers)

      The thing about the edgemark is that it does the traffic shaping to give priority to voice. (With speakeasy...) Every phone off hook costs you 90K. So a 1.544 T1 gives you 16 phones off hook simultainiously. (not 24) The balance is allocated dynamically to data. (Many systems use 64K per line) Speakeasy can bond 2 T's to give you 3MB if you need more lines.

      Behind the Edgemark, you put a standard issue 100MB switch for your network. Spekaeasy uses (used) Cisco phones which have 2 enet ports. You can daisy chain as many phones as you like and the LAST one can be a phone or a PC. We often wire each branch phone-phone-phone-workstation.

      With a SIP phone (google SIP if it is new to you) you can bring the phone anywhere in the world and plug it into a ethernet jack and you have your extension with you. No long distance etc. People just dial your local number and you can dial interoffice extensions just like usual. -coolness-

      This is a big advantage of outsourcing the virtual PBX. (or setting yours up to support WAN connections.) Sadly, while this feature is possible with Speakeasy phones, (no exaggeration...) they didn't have anyone on staff smart enough to figure out how to do it. They lied to me on several occasions and said they knew how. (but no I'm not still bitter ;-)

      With most trunking systems, each phone gets its own phone number (google "DID" it stands for 'Direct Inbound Dial' or some such) this is cool because they can bring their phones or use a softphone on a laptop.

      Why Voip?

      To me the biggest reasons to go VOIP today are to avoid the cost of a PBX or avoid the cost of long distance. Speakeasy charges about 26 bucks a month per line but since you use a virtual PBX running on their system, you have no out of pocket for the PBX. Good VOIP phones cost no more than good regular phones so that is a draw IF you are starting new or replacing equipment. But regular PBXs ain't cheap.

      If you
    • Re:Cisco (Score:2, Informative)

      by ldspartan (14035)
      Yay! I work there!

      Anyway, yes, CME (and CUE [Cisco Unity Express]) are designed specifically for this situation. It requres smart people, but so does Asterisk. And the Cisco solution has a lot more technical support than */Digium.

      Its all about choices. Want something backed by a giant corporation, and already have a Cisco router? CME. Want something Open that you can customize a /lot/? Asterisk.

      Also, check out the Cisco Integrated Services Router, and LinksysOne.

      In fact, LinksysOne is marketed at exactly th
    • 4 months with 4 people? Oh and you forgot to mention costs.
      That system just cost you at least 50k (unity alone is 15k, you need an as53xx or 54xx to terminate to pots those run 15-20k, call manager is around 10k each server... 500 phones at 350+ each).

      Anyway, maybe I'm just a fanboy, but I've deployed about 20 asterisk servers, largest being about 400 users, 4 pri's, users spread across 4 locations... $25k total, all the integration, and usability of call manager... oh yeah that deploy 2 people 2 weeks.

      the
      • Re:Cisco (Score:3, Insightful)

        by ldspartan (14035)
        I won't argue since I have an obvious bias, but Asterisk and CCM aren't really comparable. Using CCM for 400 users wouldn't be cost effective, which is why CME exists. And yes, Callmanager is about a thousand times more complex than Asterisk, and it does a hell of a lot more as well. A lot of those features probably don't matter to a lot of folks, but Callmanager runs installs with tens (and hundreds) of thousands of phones. A bit different running, say, all the phones for a major bank or credit card proces
        • Well we aren't talking about a bank, we're talking about a small software company, suggesting a 50k+ solution for even 1000 users is stupid,
          further, I personally know people who are running asterisk with 10k+ extensions, yeah you have to throw more hardware at it (10-20 servers), but not more than a CCM solution and you're throwing 2-3k pizza boxes at it instead of 10-15k HP servers...

          I know a hosted CCM provider that has 50 CCM servers and 15 Unity servers for 5000 users, yeah they have room to grow, but t