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Solutions for Small Business VoIP?
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."
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. --
http://www.asterisk.org
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.
evil is as evil does
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.
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.
Try asterisk.
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.
Yay! I work there!
/lot/? Asterisk.
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
Also, check out the Cisco Integrated Services Router, and LinksysOne.
In fact, LinksysOne is marketed at exactly this problem.
I've switched to using http://asterisk.org/ along with http://www.broadvoice.com/rates_compare.html. I think you'll find this Wiki to be a very useful resource: http://voip-info.org/
g ory&category=hardware or any other Analog Telephone Adapter (ATA), or you could use Softphones installed on employee PCs such as X-Lite (free), or similar.
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_cate
Good Luck!
http://www.gloryhoundz.com/
There are companies out there that will provide end-to-end VoIP AND the data path to do so. The one I know of, Cbeyond provides between one and three dedicated T1s along with 5-36 phone lines in CAS/PRI/Analog/VoIP format. The bandwidth is not divided per channel since the traffic is VoIP from the call switch/POTs network to the router (also provided), and therefore bandwidth not used for a call (approx. 60Kps/line) is available for data. The also have many features and other services they can provide as well, like web hosting, email, voicemail, etc. that could be cheaper bundled than purchased seperately. Also fully 911 compliant from the start, since your T1 has to have an address its installed to, and since they provide the T1(s), the routing/QOS/etc is designed specifically for call quality and rivals that of standard POTs...
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.
/ Florell_astricon_2005.html
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
Did you not see this story the other day about the new open source magazine, O3?
Their first issue "looks at reducing voice infrastructure costs with open source telephony solutions"
I suggest starting there.
Reinvent the wheel only at either a lower cost, greater effectiveness, or your own personal enrichment and satisfaction.
I just love that open NBX system so much. Heres a rant
http://www.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.
- 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:
Hope this helps.....
--Red
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.
We've installed Mitel gear in facilities ranging from Medical Clinics to Small Mom and Pop Shops. The High End Mitel 3300 would be overkill for you, but they have a small business owner flavor. We've installed both Cisco and Mitel and by far the winner is Mitel. Low maintenance, intuitive and customizeable web interface and solid performance. The small mom and pop flavor isn't that much more expensive than putting together an asterisk system and you get full support.
Switchvox http://www.switchvox.com/ will do it for you. Talk to David Podolsky there.
Email me if you have questions, I've already done the research. len at kitchenandassociates.com
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.
MoFscker
If you are going with the Cisco CME system, which is probably about as good as you're going to get in an SMB-designed VOIP system, then there is absolutely no reason to worry about physically segmenting the network. Setup VLANs on the switch and then you can plug your PC's into the back of the phones and the phones into the network jack on the wall. This method reduces wiring costs, and eases manageability.
I'm amazed asterisk@home wasn't the first thing posted here. Don't be fooled by the @Home part. This is a full fledged install of asterisk that is only limited by the hardware you install it on. You can have a working PBX in an hour. I'm planning to install this at all my remote sites (6 of them) with free extension call throughout and then plan to install it at my main location (150 phones) and have it all interconnected. A VERY powerful solution.
.iso and burn it to a CD. Boot that CD and you will get a very complete Asterisk and Linux install.
(Note: I just copied the rest of this from the handbook so I don't have to retype it all)
The Asterisk@Home project enables the home (or small office) user to quickly set up a full featured Asterisk PBX with a web based interface in about an hour on a dedicated PC. Even if you are new to Linux, Asterisk@home handles that by handling the complete Linux install for you. In order to get up and running all you need to do is download the Asterisk@Home
Asterisk@Home provides a nicely integrated install of some of the best software from the Asterisk community, such as the Asterisk Management Portal, which provides an intuitive Web GUI for configuring asterisk, and the Flash Operators Panel, which lets you see and control your Asterisk PBX in realtime, and FAX support through span-dsp.
What is included in Asterisk@Home 2.0:
Linux CentOS 4.2 - http://www.centos.org/ - CentOS is 100% compatible rebuild of the Red Hat Enterprise Linux (RHEL), in full compliance with Red Hat's redistribution requirements. CentOS 2, 3, and 4 are built from publically available open source SRPMS provided by Red Hat. CentOS conforms fully with the upstream vendor's redistribution policies and aims to be 100% binary compatible. CentOS mainly changes packages to remove upstream vendor branding and artwork. CentOS is for people who need an enterprise level operating system with stability to match without the associated cost and support.
Apache Web Server (2.0.52)
MySQL Database (4.1.12) - SQL database for Call Detail Reports and optional configuration information.
Php (4.3.9)
Asterisk 1.2 - http://www.asterisk.org/ An open source software implementation of a telephone private branch exchange (PBX). A PBX connects one or more telephones on one side to one or more telephone lines on the other side. A good example of this is a small company with 100 internal telephones sharing 20 outgoing/incoming telephone lines. A PBX can be more cost effective then having 100 direct telephone lines.
AMP 1.10.010 BETA - http://www.coalescentsystems.ca/ - Asterisk Management Panel is a web based GUI that allows you to easily manage Asterisk without having to edit sometimes complicated text configuration files. This package is can really make a difference in learning and configuring asterisk easily.
Flash Operator Panel 0.24 - http://www.asternic.org/ - Flash Operator Panel is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your PBX activity in real time. You can see what all of your extensions, trunks, and conferences are doing. You can also hang up, transfer, initate a call or create a conference call.
Festival Speech Engine version 1.96 - http://festvox.org/festival/ - Festival is a speech synthesis system. It allows you to enter text that the Asterisk@Home server "reads out loud" to anyone calling the server. Using this, you can be sure the same voice is used across the whole asterisk server.
SugarCRM with Cisco XML Services interface + Click to Dial - http://www.sugarcrm.com/crm/ - SugarCRM is designed to a be a complete customer/contact manager. Using SugarCRM we can manage all types of communications (faxes, te
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.
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
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.
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
Hemos is like...sci-fi fans;he thinks technology is cool, but he hasn't bothered to understand the science it's based on
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 just a theoretical, non-real world extreme, _as is 64Kbps_, more below.)
Regarding #1. Bandwidth, schmandwidth. It's all about LATENCY. Which is better for voice, a 50Mbps pipe or a 56Kbps pipe? Answer: Cannot tell from info provided in question. If, in the 60th second of a minute-long call, I deliver 3000Mb of voice data, I've given you the promised 50Mbps bandwidth. Unfortunately, there were 59 seconds of silence followed by an auctioneer's delightful squirt of one minute's words delivered in one second! Far better if they had been delivered less dramatically, but spaced evenly, over that minute. VOICE IS DIFFERENT FROM DATA IN THIS WAY. Had that been a big file, it wouldn't have made any difference. For file-type data, you pay your provider for the bandwidth. For voice-type data, you need to find a provider who can guarantee you evenly-spaced, regular delivery: that is, low latency and jitter. A T-1 has low latency, jitter and pkt loss; a DSL pipe may have identical _bandwidth_ but comes with no guarantee as to what is really important for voice, latency-jitter-loss.
That 56Kbps pipe? If it were a plain old $20-a-month land line from the phone company, that skimpy bandwidth would be delivering your voice with an end-to-end delay (latency) of less than 150ms; compare that to the VOIP standard (again, nominal) of 450ms. Your land line is still the Gold Standard for voice quality. (And yes, I have experienced better-sounding voice over Skype; Pure Friendly Magic! Great proof that VOIP can exceed even Carrier Grade. Someday, Vladimir, someday all the workers will have Carrier Grade VOIP.)
Regarding #2. I know that XorNand mentions overhead and is obviously aware of the following, but let's be explicit: overhead is more than trivial. You will never, never, never, never deliver voice at 64Kbps with a 64Kbps codec. That is a fake number, the limit that VOIP might approach asymptotically. Worst case? Your voice, encoded at 64Kbps, consumes about 3.5Mbps of bandwidth. (Also a fake number; we make a deal with the Devil, i.e. Delay, to keep the bandwidth down.)
The phone company standard codec, G-711, samples your voice 8000 times per second and represents the volume of your voice in that sample as an 8-bit number: 8bits*8,000 samples --> 64,000bps. The phone company then drops your voice onto the wire (on say a T-1 line) 8 bits at a time; each sample drops as soon as it's encoded, eight thousand times a second. Because this wire goes straight to the Central Office (say), the Telco does not need to add an IP address: there's only one place for it to go, the other end of the wire. Because the wire has a clocking device at both ends (the CSU that terminates a T-1) the Telco does not need to attach an RTP Timestamp to your voice: the T-1 circuit does that too. Because the voice samples can't leapfrog eachother in the wire, or get lost, the Telco does not need to attach a TCP sequence number or acknowledgement; the CSUs know whether a sample is to be used as voice or data, and handle multiplexing, so there is no need for a TCP/UDP port number.
You can see where this is going, right? VOIP takes the same sample, and to deliver it attaches an RTP header for timing/sequencing/codec info, a UDP header for port number, an IP header for end-to-end addressing, and an Ethernet header to get you across your LAN. That 1-byte sample is now dozens of bytes long. It's as if to carry 8000 commuters to work you sent out 8000 trains, each with a string of locomotives to pull a single commuter down the rails.
Disclosure: I work for a company that provides voip backbone. Which one doesn't matter, the experiences are universal.
If you're going to try voip, be sure to give the providers full details about what you want and what you want from them. Don't let the sales types snowball you; make sure that you get details and hard information. And make sure *you* do the same with them.
I'm supporting voip, and we generally run into three general problems: People with unrealistic quality expectations, Bad PBX setups, and lack of information.
Quality: People, POTS has been around for a long time. VOIP has been around for less than a decade. There are going to be mistakes. Everyone - everyone - is still figuring this stuff out. In five years, probably less, this will be seamless and easy; but if you're looking to do this to save money now, please realize that there will be some bumps along the road.
Bad PBX setups: This has been our #1 headache in customer satisfaction. VOIP gets set up, and the customer begins calling in with a variety of complaints. Examination of the logs shows us that *our* call handling is working right. We have to go to the customer and tell him that we suspect his PBX setup is probably bad. The customer is quite rightly suspicious - everything worked fine pre-voip, why should it be a problem now? The answer is that voip is not as robust as POTS; the customer PBX HAS to be set up to properly handle and deliver ISDN message traffic between the PBX and the gateway, or you get problems. So the customer has to call their PBX vendor, and pay them a lot of money, to come out and look at the setup - and the vendor may well not understand what is needed and point the finger back at us. This can go on for a while...
My reccomendataion: If you want a PBX system (and really, it's a good idea, voip or no - a PBX is a server designed to route phone traffic, and does a good job of it) then just go with asterix. A vendor who supports asterix is far more likely to be technically conversant with the needs of voip, and it will be a lot eaiser to take over management of the box internally than a traditional PBX. (Ever tried to program a nortel PBX? Eeeshhh...)
Lack of information: I spend weeks, sometimes months troubleshooting the tough cases. I ask specific questions, get answers, and a month or two later, find out that they omitted a trivial detail - that is the cause of the whole problem. I can't be too specific, but GIVE YOUR PROVIDER FULL DETAILS when they ask for them. If you don't know, say so...admitting ignorance now is a lot less painful than burning 40+ man-hours of your time and mine before doing it.
Hope this helps.