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Solutions for Small Business VoIP?

Cliff posted more than 8 years ago | from the from-copper-to-network dept.

Communications 232

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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232 comments

VoIP Privacy (-1, Offtopic)

m3lt (933405) | more than 8 years ago | (#14143683)

But I think phone calls should be private, and the only way for a police department or FBI to wiretap should be with a court order. There should be hoops to jump through, and it should not be easy to do. But maybe there is more to it? Congress gave telephone companies $500 million to buy new equipment to comply with CALEA. Why should Internet companies not receive the same treatment? Is it because Verizon, SBC and the other former Bells have well-connected lobbying outposts in Washington, D.C.--but Vonage, 8x8 and other VoIP start-ups do not? According to the article, congress gave telcom companies $500,000,000 to enforce the laws they passed? Why doesn't the government give me money to enforce their pollution laws, so I can get my car fixed up. Instead I have to pay to comply with the law. People must be aware they are giving something up here. They are giving away freedom. What if some day comes, when a David Duke wins the white house? Congress is filled with people who vote along lobbyist lines. And we end up with laws that remove our consitutional rights- like having police wiretap without a warrent or snoop around the library to see what we are reading. What if they take away our 2nd amendment rights, first by requiring registration, than banning assult style wepons, then slowly, state by state, taking away wepons you already own. What if the states decide to put up a camera on every street corner.... then one day in your house. The point is the founding fathers did not add the Bill or Rights because it sounded like a nice set of rights. They added those Rights so the people could fight an overbearing government if the need ever came. What if England had decided the colony could not have any guns, and decided that neighbors must report what other neighbors say. We would not be a country today, we would be English. The founding fathers gave people certain Rights to make sure we stay free. Those that give away those Rights are comminting suicide for the rest of us. They are chaining us all. Rossoue was right "Man is born free, yet everywhere he is in chains". People, don't give you your rights!

My experience (4, Informative)

2.7182 (819680) | more than 8 years ago | (#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. --

i just fucked your momma (-1, Troll)

Anonymous Coward | more than 8 years ago | (#14143741)

see subject

Re:My experience (4, Informative)

XorNand (517466) | more than 8 years ago | (#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.

PARENT IS A NIGGER! MOD DOWN! NIGGER ALERT! (-1, Troll)

Anonymous Coward | more than 8 years ago | (#14143889)

Stupid fucking nigger!

Re:My experience (2, Insightful)

NoMoreNicksLeft (516230) | more than 8 years ago | (#14144047)

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 clever enough to know the guy is asking about asterisk... that part is probably obvious to him. He's more concerned with how to manage a network, bandwidth and hardware wise, to use this. With maybe a little "what's the best VOIP phone for your money" thrown in.

Asterisk (1, Informative)

fiber0pti (728998) | more than 8 years ago | (#14143686)

http://www.asterisk.org

Re: Asterisk (2, Insightful)

Py to the Wiz (905662) | more than 8 years ago | (#14143709)

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 overkill for small businesses, as well as for any but the most elaborate homes. (Should be doable for a few hundred dollars, I'd guess.)

Or am I just missing that someone is selling such a beast already?

+1, Baited question (0, Troll)

bradleyland (798918) | more than 8 years ago | (#14143824)

Screw the negative karma, anyone who reads Slashdot knows that 90% of the answers are going to be *.

Re:Asterisk (4, Informative)

amliebsch (724858) | more than 8 years ago | (#14143825)

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 (3, Interesting)

e4g4 (533831) | more than 8 years ago | (#14143848)

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, software, hardware and other choices - as well as finding out the quirks and perks of each and mountains of setup info is the voip wiki [voip-info.org] .

Cheers, and good luck - you may need some in the process.

Switchvox! (0)

Anonymous Coward | more than 8 years ago | (#14143866)

I switched my company to switchvox from Avaya and haven't looked back. We're saving a few hundred dollars a month in phone calls and since switchvox is built off of asterisk it has all of the features of a modern PBX (even some features they added themselves).

My latest favorite feature from them is a firefox plug in that lets anyone right-click on a phone number and dial it. It rings your extensions and automatically connects you with the number. No waiting and no errors when dialing.

Has anyone else used their solutions? It cost about $2k to setup my office, but everyone loves it.

The obvious choice. (4, Informative)

killjoe (766577) | more than 8 years ago | (#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.

Re:The obvious choice. (3, Informative)

kasparov (105041) | more than 8 years ago | (#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 (0)

Anonymous Coward | more than 8 years ago | (#14143688)

looks like its voip night

Simple (1, Redundant)

KiranWolf (635591) | more than 8 years ago | (#14143692)

Asterisk [asterisk.org] .

Re:Simple -- The great thing about Asterisk (1)

ursuspacificus (769889) | more than 8 years ago | (#14143900)

So... the great thing about Asterisk is you don't have to use it only for VOIP connectivity to the outside world. You can connect out using Full T1 Data, 24-channel T1, Virtually any number of individual POTS lines or VoIP (by way of SIP, MGCP, IAX or other protocols)... That way you can grow your system, migrate to different connectivity technologies, be free from vendor lock-in for phone hardware and get on with your life.

I'm in the process of doing a migration off of Broadvox and onto an Asterisk-based system. We have about 30 users in 2 offices. I love the flexibility that Asterisk offers. I'm putting 1 Asterisk PBX in each office with a block of 100 DIDs. Because Asterisk is hardware agnostic, it works with just about any "standards based" VoIP phone. That covers the local traffic. Connectivity to the outside is being done through an outfit called Junction Networks over Asterisk's IAX2 protocol. It's a killer deal. If we choose to dump our connectivity provider and revert to POTS, we can still use all of the gear we already have... just plug a card or 2 into the Asterisk PBX, tweak the dialplan and off we go.

BTW, I use Asterisk at home, too....

I am totally sold!

Hope you don't get robbed (-1, Troll)

Anonymous Coward | more than 8 years ago | (#14143693)

or have a heart attack, because Voip doesn't support 9-1-1!! If only klerck didn't have VOIP when he was attacked by a shotgun wielding negro.

VoIP is not cheaper (3, Interesting)

Py to the Wiz (905662) | more than 8 years ago | (#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).

Re:VoIP is not cheaper (1)

fiber0pti (728998) | more than 8 years ago | (#14143705)

That's a load of bull. VoIP is extremely cheaper for small businesses. I have consistantly saved customers over $500 a month on their phone bill with only a ~$5,000 entry fee. You can't beat that ROI.

Re:VoIP is not cheaper (1)

Py to the Wiz (905662) | more than 8 years ago | (#14143721)

Well, in your case, it very well may be cheaper. I'm just saying don't automatically assume that VoIP will be cheaper just because it's VoIP. In our situation it very clearly was the more expensive option. As always, YMMV.

Really cheap small-office pbx (1)

Nick Driver (238034) | more than 8 years ago | (#14143776)

For very small offices (up to 32 phones), take a look at TalkSwitch small PBX [talkswitch.com] . Prices start around $700 for the entry-level 4 local extension POTS unit. The 8-extension unit with 4 VoIP ports is $1800.

Depends on your provider... (1, Informative)

Anonymous Coward | more than 8 years ago | (#14143783)

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 [cbeyond.com] 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...

Re:VoIP is not cheaper (2, Interesting)

Trejkaz (615352) | more than 8 years ago | (#14143989)

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.

This is not necessarily true (1)

theschwartz (766807) | more than 8 years ago | (#14144038)

My company is offering VoIP bundles for 4 to 24 users, and include the handsets, POE switches, router, gateway, and with certain systems, even Wireless Access Points. System PDF [alliedtelesyn.com] Here is a writeup from crn.com crn.com [crn.com] Info on the individual products in the systems can be found here Handsets [alliedtelesyn.com] Gateway [alliedtelesyn.com] 8 Port POE Switch [alliedtelesyn.com] 24 Port POE Switch [alliedtelesyn.com] 24 Port Layer 3 Switch [alliedtelesyn.com] Router [alliedtelesyn.com] WAP [alliedtelesyn.com]

Astrisk (1)

Akash (442677) | more than 8 years ago | (#14143701)

asterisk should do good.. get that and the asterisk at home live disk and the digdum card depending on how many pots likes you have.. i dont know if you want to go all voip or still have the voip to pots conversion happen at the pbx

Design your own (-1)

Anonymous Coward | more than 8 years ago | (#14143719)

VoIP isn't anything complicated. ( I have designed several such systems myself and I know what I'm talking about) Hire even a single half decent coder and you will have a system up within a day or two. The source for Skype isn't available, but it's a no brainer really, you just need to shovel a few packets from ADC to DAC remotely in duplex. Keeping an address book is the harder part, and handling the sign on/off mechanism - but if you are a small company you're looking at a few hundred users most and point to point. All the compression and buffering problems normally associated with VoIP are fairly moot in a broadband network, Skype/Vontage etc only retains that to work well with 56k dialups.

Moderation gone mad!! (1)

ldspartan (14035) | more than 8 years ago | (#14143787)

+1 Informative?! Gah! Mods! Stop falling for the clever troll!

"but it's a no brainer really, you just need to shovel a few packets from ADC to DAC remotely in duplex. Keeping an address book is the harder part, and handling the sign on/off mechanism"

What are you kidding? An address book harder than dealing with jitter? Pay attention, moderators!

Re:Moderation gone mad!! (0)

Anonymous Coward | more than 8 years ago | (#14143943)

Troll? No. Clever? Thanks, yes I know, but don't let that frighten you.
As an AC I don't get any real chance to defend my side but I can
promise you that I have built at least three _working_ systems.
Jitter can be solved many ways, simple reordering, windowed backbuffering,
lots of fancy stuff - or for a budget DIY system just don't bother about
it too much, it's not like the OP asked for a fully commercial system is it?
Here's a few obvious links to help. I'm guessing you are enraged by my suggestion
because you work for a commercial VoIP provider. What can I say? Please grow up.
I also guess you have never actually built such such a system either, try it, go on,
fire up that C compiler and amaze yourself at how easy it is. Really, you don't need a load of fancy stuff, VoIP hardly even needs a processor, man you could get a 4MHz Z80 to
do most of what is required.

here [thefreecountry.com]
here [linuxjournal.com]
here [openh323.org]
here [linuxdevices.com]
here [voip-info.org]
and here [datacompression.info]

similiar position (5, Informative)

sgeye (757198) | more than 8 years ago | (#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.

Re:similiar position (1)

pavera (320634) | more than 8 years ago | (#14143859)

Did you look at any asterisk resellers?
I'm one :) we can beat any price cisco can give you, and we support our solutions 100%
http://www.singlepointnetworks.com/ [singlepointnetworks.com]

Re:similiar position (1)

sgeye (757198) | more than 8 years ago | (#14143921)

We didn't get an asterick quote, unfortunately I seem to be the only proponent of open source in our department, everyone else is scared of the support issues. I will send you a quote request and put it on the table though.

We use the Cisco IP Phones & Service.. (5, Interesting)

PogiTalonX (449644) | more than 8 years ago | (#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.

Re:We use the Cisco IP Phones & Service.. (1)

Etyenne (4915) | more than 8 years ago | (#14143904)

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.

Re:We use the Cisco IP Phones & Service.. (1)

ldspartan (14035) | more than 8 years ago | (#14143934)

The Cisco IP phones support SIP (i.e., a whole bunch of third party IP PBXen, including asterisk) and the Cisco propriertary signalling protocol (SCCP). In SCCP mode, connected to Cisco Callmanager, they can encrypt both media and signalling. In SIP mode, they can encrypt neither.

There's no good solution for SIP crypto at this point, due specifically to its interoperability needs.

Re:We use the Cisco IP Phones & Service.. (1)

Tmack (593755) | more than 8 years ago | (#14143965)

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, it will always appear as the main office number. The "plugging it in half way across the country and still dial by extension" part is of course not possible with legacy digital services without a VoIP gateway of some sort, and your configuration probably involvs a VPN for the phone to connect back to your company's lan to place the calls (but yeh, it is neat).

YIAATE (yes I am a telecom engineer ... for a VoIP provider ... but one that supplies its own transport rather than rely on the intarnet)

tm

just wait for the next BIG thing: wifi/cell headsets that handoff between both...and to your landline (http://www.xchangemag.com/articles/561air4.html [xchangemag.com] ).

Put them on their own network segment... (1)

bergeron76 (176351) | more than 8 years ago | (#14143736)

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.

Re:Put them on their own network segment... (1)

lazybeam (162300) | more than 8 years ago | (#14143798)

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

Just get a bunch of PAP2s and normal phones. :)

Re:Put them on their own network segment... (1)

bergeron76 (176351) | more than 8 years ago | (#14143970)

I'm not sure what a PAP2 is. This tech is somewhat new, so please forgive my naivete...

Cisco (1)

huber (723453) | more than 8 years ago | (#14143740)

It may be a closed Sysytem and piss off a few slashdotter, but Callmanager is a great system. Callmanager express is resonably priced. Its flexable, scaleable and overall works very well. We Just deployed aroung 500 phones using two call manager servers and a single unity messaging server integrated with our lotus domino system. Total time from equipment delivery to deployment was 4 months with 4 people.

Re:Cisco (2, Informative)

ldspartan (14035) | more than 8 years ago | (#14143768)

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 this problem.

Re:Cisco (1)

pavera (320634) | more than 8 years ago | (#14143814)

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 second person used to work for cisco in TAC on the voice team (supporting call manager, and call manager express), avoid those products like the plague. They are a pain to setup, once they're working you have to babysit them, oh yeah and unity requires that you use exchange server, so have fun with that. TAC won't even talk to you or support your unity install unless you have an MCSE to talk to them, and it helps if you have at least one person who is CCIE voice... Also, its just plain expensive and doesn't do anything extra that asterisk does.

Re:Cisco (3, Insightful)

ldspartan (14035) | more than 8 years ago | (#14143858)

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 processing house than running 400 phones in a small or medium sized business.

Different strokes for different folks, but you'd be stupid to dismiss either option out of hand.

Re:Cisco (1)

pavera (320634) | more than 8 years ago | (#14143905)

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 they are at about 75% capacity right now on those servers. That's not to mention the 10 as5400's that they have... all together their infrastructure cost more than 4 million and they pay 200k+ each year in service contracts. But sure I guess if you're just spending someone else's money CCM is a good way to go.

Anyway, CME is still more expensive than asterisk solutions I've priced, and then you're stuck with very limited expandability.

Re:Cisco (2, Interesting)

ldspartan (14035) | more than 8 years ago | (#14144009)

"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"

As is dismissing a solution out of hand thinking hard about it.

"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 they are at about 75% capacity right now on those servers."

I can't imagine how they achieved such terrible density. That many CCMs should be supporting phones numbered in the tens of thousands, minimum. Unless they did something stupid, like buying 7815s. Then again, I'm on the development, not support side of things.

"Anyway, CME is still more expensive than asterisk solutions I've priced, and then you're stuck with very limited expandability."

Of course it is, since you can use $50 budgetone handsets with Asterisk, and need to be buying at least 7905s with CME. And you're getting the call-processing machine for cheap/free. And Cisco TAC is going to be at least a bit sad when you call up with voice quality problems and don't have QoS all over your network, or have no idea what you're doing. But I do believe you get something for your money.

Certainly, if you assume that you're in an environment where management is unlikely to approve a home-grown / not-supported-by-someone-big solution, Cisco is the best thing going.

LinksysOne is exciting to me. Its targeted exactly at this situation (well, maybe a bit more small law office than small software company...) and looks to offload as much administration as possible, and move those cost centers that bug you so much (as5400s, etc.) to the ISP side.

Re:Cisco (1)

huber (723453) | more than 8 years ago | (#14143871)

While some of your comments make sense. Some dont. We run three pri's and get them at a very good price. did's are cheap as hell. Unity integrated just fine with Lotus Domino which is what i said in my original post. Deployment took as long due to manually unpackeging the phones and bringing then to the physical locations spread out over 8 facilities. I have never had a problem with TAC. they are very responsive. The more i think about it, maybe your post is flaimbait?

Re:Cisco (0)

Anonymous Coward | more than 8 years ago | (#14143896)

I would have to agree. Supporting 14 offices on Cisco call manager, I have never had a problem with Cisco TAC. They have been able to answer all my questions that I haven't been able to find on their site. You don't need to be a CCVP or MSCE to get support.

Re:Cisco (1)

AgentScummy (927127) | more than 8 years ago | (#14143897)

I run a full Cisco Callamanger system by myself. I set it up by myself and I have never had to call TAC (because they are worthless for the most part) I have clustered Callamagers, Unity, Personal Assistant and use a 2651X router for the gateway which I might I add we use to bridge SIP calls from our Voxeo IVR and SIP proxies. My only gripe is the constant patches from Microsoft. It;s nice too that I can manage it from anywhere in the world provide I have an internet connection.

make sure you get a 2nd line! (2, Informative)

lkcl (517947) | more than 8 years ago | (#14143747)

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.

Re:make sure you get a 2nd line! (2, Informative)

lazybeam (162300) | more than 8 years ago | (#14143790)

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).

details / explanation... (1)

lkcl (517947) | more than 8 years ago | (#14143817)

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 ports... ... and your ISP has set the MTU _so_ high that it makes absolutely bugger-all difference: all those "internet surfing" packets come in on an MTU of 1500 which _totally_ dominates your line for so long that the UDP SIP/RTP packets don't stand a chance.

hence the requirement to have a second _separate_ line, on which _nothing_ else comes in.

regarding putting the public IP address direct on the box [there are other ways to achieve that other than doing an ADSL card in the server, i knowwwww - it's just that the kit is expensive, and ADSL PCI cards like the bewan - unicorn chipset - and conexant falcon 2p cards - are £12 to £25].

what you can do there is write a custom firewall that copes properly with the setup - and the problems associated with SIP/RTP behind NAT can be made to vanish by having asterisk actually on the same box that's doing the outgoing routing.

the other advantage of having asterisk - even though it's a complete BASTARD to set up - is that it provides a common interface for all those incompatible SIP phones your company is about to buy because they won't listen to advice about making sure you only buy the same make, model and brand of SIP phone for _evverryone_.

SIP is a bastard protocol and no two SIP phones - hardware or software - are properly interoperable.

asterisk helps take some of the non-interoperability out of the equation, but not completely.

Re:make sure you get a 2nd line! (1)

lowlands (463021) | more than 8 years ago | (#14143939)

It's a very bad idea to use your Asterisk server as the ADSL gateway. Asterisk is time sensitive and you want that box to do Asterisk and nothing else. A small delay in trying to write some data to the harddisk can already cause hickups in people's conversations. And you really do not want to be support when that happens. People get furious if their phones calls suck in terms of (their perceived) quality. Obviously if you put the Asterisk box behind the (ADSL) gateway than you may get NAT issues if you use SIP. If you can get multiple static IP addresses (one for the ADSL gateway and one for the Asterisk box) than I would definitely do that. Best is to avoid SIP and use IAX to connect to your ITSP.

My recommendation is to use Asterisk/VoIP internally (so phones and server) and hook up a T1 or PRI (ISDN Primary Rate with 24 channels) to your preferred phone company. Get some "pay as you go" service from an ITSP and if the quality through your ITSP is good, route outgoing calls first to the ITSP and if they fail fall back to the T1 or PRI. Also get a block of DIDs for all employees so they can be reached directly, offloading the receptionist.

Asterisk has quite a steep learning curve. If you want to get up and running asap then hire a consultant with good references and shell out the cash. You will end up with a well working system and can start to learn Asterisk from there on. Also get the two Asterisk books, read everything you can find at voip-info.org and subscribe to the Asterisk mailing lists. Good luck!

Re:make sure you get a 2nd line! (1)

Wesley Felter (138342) | more than 8 years ago | (#14144002)

By my calculations, a 1500 byte MTU should only cause 10ms of jitter at 1.5Mbps, which doesn't seem too bad.

If you're going to get a separate line for voice, you might as well get a PRI and a VoIP PBX on your premises, which would eliminate Internet problems altogether.

NAT should not be a problem in a business environment if you just don't use NAT.

As for GSM codecs, I wonder if employees would enjoy cellular quality office phones.

As well as 2'nd provider (1)

WindBourne (631190) | more than 8 years ago | (#14144059)

Look, if you are using a 2'nd broadband for reliability, then you might as well back up the other part of that; the voip provider. I Did a few asterisk installs, and saw them burned by one company (not only did they not handle the rush, but they did not handle their support well; ignored calls too often).

I work for a co. that installs VOIP systems (1)

jred (111898) | more than 8 years ago | (#14143748)

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/ the phone sys using a PRI (?) to the phone co, AND you have managed switches, you should be ok".

asterisk (2, Informative)

max born (739948) | more than 8 years ago | (#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.

Re:asterisk (1)

plierhead (570797) | more than 8 years ago | (#14143911)

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 take you a heck of a long time to get on top of. Count on a fun but steep learning curve on concepts like the asterisk config files, VOIP protocols, flashing your IP phones to get their firmware to the right level, etc.

If you really want to truly master your phone system, and be able to do whatever you want with it (hey, we're all geeks here, right?) , then go asterisk, and just do it all yourself. Make your PABX a corporate asset that you have hack, extend and exploit.

But if your primary goal is to get the office phones up and running with minimal cost (in both time AND money) then finding someone else who knows asterisk and VOIP generally will likely be way cheaper.

Asterisk is the go! (0)

Anonymous Coward | more than 8 years ago | (#14143767)

I have used it internally as a PABX replacement and it worked a treat.

If you can - install FXO ports as a backup - and offload your voice traffic to an outside provider. That way you don't have to worry about setups of zones etc... as they will look after it.

The design that I proposed here to replace our pabx was:

SPA-3000's in remote branches (allows FXO and FXS ports)
Asterisk (with its own UPS & Internet link) talking to an outside provider
a few PXS ports to allow for those "I like my analogue phone" people
Ethernet-based Voip Phones for those desktops that need them
VoiP-USB phones + Xlite/OpenH323 for notebooks (so they can use them travelling too).
Connection to FWD is always a good idea too... I know it was very useful here in Australia to get to some USA 1800 numbers..

Best advice:
Keep your extensions.conf file as "uncomplicated" as possible.

BYOD @ Broadvoice (5, Informative)

TheRealFritz (931415) | more than 8 years ago | (#14143782)

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]

Re:BYOD @ Broadvoice (1)

jedaustin (52181) | more than 8 years ago | (#14143940)

I made the mistake of using them awhile back.. First two months went great, on the third month it totally went to crap. I recommend having multiple voip providers especially if one of them is Broadvoice.

Re:BYOD @ Broadvoice (1)

the.house (917996) | more than 8 years ago | (#14144007)

1. build a beowulf cluster of wrt54g routers running asterisk on openWRT... 2. ? 3. Profit!

Voip (1)

clifffton (912293) | more than 8 years ago | (#14143793)

We have a 200+ handset 3com MAC (I won't call them VoIP, 'cause they aren't) at work and have been very happy with the system. It was 30% cheaper than 2 Cisco quotes I got, and I was frankly scared of CallManager since it ran on Win2K. It's a decent system, and the new handsets have SIP available. Management is all web-based and fairly easy once you get the telco terminology figured out. I do suggest whatever you do allow for a seperate network segment or a VLAN for best performance.

Small Business Voip Implementation (2, Insightful)

ravenvijesh (934885) | more than 8 years ago | (#14143794)

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 company yet. There are always call quality issues unless you manage to get physically near your termination provider and you have a fat pipe from your offices.

The fact that you're a part of a development house is going to help out a lot when customizing your solution. Asterisk really isn't that complicated - modifying it so that it fits your companies needs and provides true business benefit is probably the biggest thing. (Like integrating it with your existing CRM solution or back ending VoIP to your database).

There is a PDF which helps on the overall analysis and how Asterisk can be pretty usefull for smaller businesses =
A Voip Small to Medium Business Analysis [skyyconsulting.com]

commercial system (1)

arhub (934887) | more than 8 years ago | (#14143800)

It sounds like you will want to use some kind of commerical system I have used asterisk and its a good product, but for a fair sized phone system you will want to go with either a Cisco or 3com solution. The company I work at sells and installs 3com phone systems. The 3com systems are relitivly simple to install and can be easily expanded to handle as many users as you need, you can choose to use regular analog lines or a T1 trunk for you incoming phone lines. Although I work with 3com systems all the time and prefer them I would encourage you to look at both Cisco and 3com and choose a system that suites your needs.

Find a consultant.... (1)

mewyn (663989) | more than 8 years ago | (#14143802)

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.

Re:Find a consultant.... (1)

ian_mackereth (889101) | more than 8 years ago | (#14143985)

Wander into any library and look for the 14 year old with the band-aid holding his thick-frames glasses together.

That's who you want installing your VOIP software!

Asterisk has saved us over $1 million in the ... (5, Informative)

mflorell (546944) | more than 8 years ago | (#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]

EVIL... (0)

Anonymous Coward | more than 8 years ago | (#14143997)

I read your presentation. How do you spell "evil"?

I spell it "predictive dialling"

Don't forget about the network (3, Insightful)

g-san (93038) | more than 8 years ago | (#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.

IPT tips (1, Insightful)

Anonymous Coward | more than 8 years ago | (#14143815)

I have implemented many offices with Cisco Call Manager, sorry as this is the only VoIP PBX I have experience with.
The best tip I can give is to make sure that you have a good infrastructure in place, that supports QOS (quality of service). Typically I only work with Cisco equipment, but I can say that this is an important step. This helps preserve needed bandwidth for call setup and for current active calls. Without it calls will drop, and you will experience choppy / robotic conversations. A Cisco 3560 access switch would be a good choice, as it allows you to use auto qos, and modify the qos statements to fit your needs.
You also might want to look into equipment that can do SRST (survivable remote site telephony) if the IPT (IP telephony) equipment isn't in the same office as the phones. This will allow you to do basic phone service for the office (make and receive calls) with out the need of the CCM (Cisco call manager).
In short, not sure if Cisco Call Manager is a good solution for the size of your company, but since you own /rent a PBX it cannot hurt. There are many consulting firms out there that can host the call manager in their own NOC, making it an attractive solution.
Hope this helps ~MP

Is this a trick question? (1)

zaphodb001 (527707) | more than 8 years ago | (#14143883)

I have to ask the obvious question here, WHY, you don't tell us WHY? VoIP = Doesnt "MEAN" anything to most people besides "Cheaper". So is that what you are after? Shifting costs around using Asterisk is fine, playing it safe and buying a commercial product is fine. But the real question is WHY? What are your calling patterns? Some small businesses will be better off handing out cell phones and doing bulk purchase of in/out bound minutes over a fixed configuration solution that will require training. So, before all the techno tower of babble.... Answer the question -- WHY? Cheaper? More Applications? Get you closure to your customers? Make you more Competitive? Voice is just another application in the data center to be managed based on a clear vision and determinable ROI.....

Did you not see... (2, Informative)

syukton (256348) | more than 8 years ago | (#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.

SBC and Cisco are no fun at all... (0)

Anonymous Coward | more than 8 years ago | (#14143925)

I just had the unfortunate experience of having an SBC installed Cisco VoIP jammed down my throat. The Cisco software is far from intuitive, it might make more sense if you're a telecom guy, but I am not.

Lots of nice features like a company phonebook that will "update" when you make changes are not that easy. You have to dig and find the three places you must change all of the data, so everything will work right.

911 thinks that we are a nursing home (We are not). SBC still has not fixed this problem no matter how much I have yelled at them about it. Working in an industry where employees must call 911 everyday about car accidents makes this fact even more fun. On top of that SBC claims we can only have 1 number that gets reported to 911 no matter where you call from in our massive facility. Even though we were told this was not the case before the system went in.

Caller ID will display the proper number of the phone you're calling from (After I had to get an SBC tech to fix how our system was set up) but 911 won't work? You have to love legacy systems that SBC won't update.

What else... Oh the lead designer from SBC who put in our system never gave me any of the passwords to all of the routers he set up, so it was fun yelling at SBC for a month to get those.

Backup old school phone lines... I won't even get started on those. I will leave it at they don't work.

I was not involved with the design from the start, but even if I was there is a bunch of stuff I would have assumed these systems would have that they don't. If you want your system to last make sure you talk to someone who has used at least a couple of these systems before.

Have fun!

comercially supported FOSS (1)

buddha42 (539539) | more than 8 years ago | (#14143946)

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.

Re:comercially supported FOSS (1)

Lost+Found (844289) | more than 8 years ago | (#14144063)

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.

Asterisk may have messy code, but in my experience it's stable and it will smack the shit out of proprietary alternatives in terms of call rates, etc.

ShoreTel (0)

Anonymous Coward | more than 8 years ago | (#14143956)

We use a ShoreTel system works wonders you can have a pots or pri feed it. Real easy to manage also.

Zultys (0)

Anonymous Coward | more than 8 years ago | (#14143974)

We're running a Zultys [zultys.com] system with good results. It's a SIP based system with lots of auto attendant, voicemail, fax client, and other feeatures and it's cheaper than Cisco! You can use SIP phones from Zultys, Cisco, Grandstream, Polycom, etc. I personally am reluctant to run VoIP/firewall/router/kitchen sink all on one of the Cisco ISRs, even though we have one of those already.

Million dollar mistake (0)

Anonymous Coward | more than 8 years ago | (#14143984)

Whatever you do, do not use CIC from I3. After spending nearly a million dollars on it we are now migrating to Asterisk. Only thing we did right was use Cisco hardware for the PRI gateways.

Asterisk (2, Interesting)

Denis Lemire (27713) | more than 8 years ago | (#14143990)

I handle the IT for an Edmonton based WISP. When we moved offices almost a year ago we left our old Centrex system behind and built our own PBX using Asterisk. Overall we are happy with the setup, though it has a learning curve.

Once you resolve all the issues with echo cancellation, you'll end up with a very flexible setup. Best of all, because of its open standard nature you will not be marrried to any particular vendor of handsets.

It takes a little bit of work to get everything running to the spec you're looking for, but the results I would say are well worth it.

VOIP system (0)

Anonymous Coward | more than 8 years ago | (#14144006)

I have had some experience with the system from TalkSwitch. The system allows VOIP lines and analog lines. So far it has worked well except for some minor issues, mostly to do with configuration problems. We run a Talkswitch system CVA with the airway cordless system for the actual handsets. The TalkSwitch is defiantly worth checking out as an option and the cordless phones means I didn't have to worry about wiring. Talkswitch: www.talkswitch.com Airway phones: www.homewireless.com

Why bother? (0)

Anonymous Coward | more than 8 years ago | (#14144014)

I have to say why bother? With the price of voice T1s and the price per minute of long distance you get with them I just don't see what the fuss is about. We bougnt a Mitel SX200-ICP about 2 years ago, all the phones in the office are IP and connect into POE switches. We can also take phones home and connect into the phone system. But rely on Vonage or another provider to make outbound calls? Forget it.

Experience with Vonage (1)

pointyhairedmba (698579) | more than 8 years ago | (#14144045)

I recently started a small 4 person startup in a shared office space with another 5 person startup. I decided to try Vonage based on cost grounds compared to SBC's small business service. We also have a commercial cable modem service. Unfortunately we had to cancel the service after a couple of months. The call quality became unusable anytime more than 4 or 5 people were using the internet at the same time. Now I'm not a super tech person (but am comfortable with hardware/software), and I was looking for a simple, cheap solution. Vonage was unfortunately not at the quality of service we needed at this time. I suspect that it would work just fine for home use though.

e-mail me. (1)

numbski (515011) | more than 8 years ago | (#14144050)

I'm serious. I've been doing this day in and day out for months now. The whole thing comes out costing about 1/4 of what a commercial solution is, the quality is better, and you can do insanely cool things with it. :)

It's just more typing than I'm willing to do right this second. The name of my company is oss|solutions.

My brother does this stuff regularly (1)

bill_kress (99356) | more than 8 years ago | (#14144051)

You might check out his podcast where he mentions it. If you email him with questions about it, he will be guaranteed to add it to his podcast.

The name is Doug's daily tech--it's about his job as an IT manager and has some good insight.

Since he is currently upgrading his company to asterisk, I'm sure he'd love to discuss it.

He made a custom Knoppix distro with Asterisk and some other utilities needed to run such a beast. Send an email asking if you are interested: ddtcast@gmail.com.

http://wiki.ddtcast.com/wakka.php?wakka=HomePage [ddtcast.com]

Speakeasy (1)

Jeffrey Baker (6191) | more than 8 years ago | (#14144052)

Don't implement this yourself. Call up Speakeasy. They will set you up with the phones (or you can buy them yourself) and will configure, host, and operate the service. The price is very low and I haven't had the first problem with the service. It's 1000 times better and 100 times less expensive than my old Lucent PBX with WorldCom T1 service.

What luck... (1)

phpsocialclub (575460) | more than 8 years ago | (#14144053)

I have been researching this very topic all day, and now this is in /.

We are looking at replacing our existing Nortal Merdian system with a Nortel BCM 400, so we can keep all of our existing desk phones. The Nortel BCM is about 5k with 32 extensitons and 4 voip phones and licenses. We will use a the voip phones at our remote office untill it out grows and then add a BCM 50 locally and bridge the two offices over VOIP.

Does anyone have any comments on this set up. I would love to get astericks in the mix, but I am not that smart :) and I will not be around the main office to admin it.

I also found this intereting link, about OpenVPN and astericks with 2 T1s

http://www.softwink.com/papers/Installation_Securi ng_VoIP_With_Linux/ [softwink.com]

I would love to put the astericks system in the remote office instead of the BCM 50.

Recommendation: Asterisk @ Home (2, Informative)

RedLeg (22564) | more than 8 years ago | (#14144062)

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

9 months with Asterisk (1)

TBC (11250) | more than 8 years ago | (#14144074)

We have been using Asterisk for about 9 months. We came from an Altigen system. Our configuration was:

Digium 4 port T1 card and ADIT Channel bank with 8 FXO & 16 FXS ports
Cisco 7960 SIP Phone
Generic selection of SIP/IAX phones
Intel Server Class hardware (ECC Memory, RAID, etc.)

The Altigen system was a 8x16 system, and had a really good call queue system. We just needed more extensions. My goal was to duplicate the capabilities of that system. I started using AMP, the asterisk management portal as my configuration "GUI". In our office, I have 2 Linux/Unix people, and 20 windows techs, so my goal was a user friendly management system that I didn't have to baby sit. Unfortunately, when we started, AMP didn't support call queues. I hand-coded in the queues, and had a problem with queue calls dropping directly to voicemail. I'm in the process of transferring all of my extensions into the latest version of AMP, but I still have a few issues.

We have a number of issues that I believe will be fixed when we switch out the config files, but as it is right now, Asterisk is very unforgiving of errors in the dial-plan configuration files. If I had the option to do it over again, I probably wouldn't have gone with Asterisk. I still have problems where a "ZAP" or analog extension will simply "lock up." I have an issue where SIP calls will unpredictably fail until the extension re-registers. We have set up a connection with voicepulse to do outbound long distance, and it's OK as long as traffic isn't too heavy.

My advice is to consider Asterisk under the following conditions:

You need a VERY simple phone system. An Asterisk server with 4 FXO lines, 8 VoIP extension, and simple IVR menues to get to the extensions.

or

You are looking for a complex phone system, and can dedicate the time to hand-create the dial-plan files to be exactly what you need.

or

You can pay Digium or a consultant to customize the phone system exactly for your needs.

Asterisk has so many capabilities, but (not to knock the developers) it is too easy to crash the engine with a misplaced dial-plan entry. I created a "time-and-temp" application just for fun. It's absolutely amazing what you can do with it. Unfortunately, it isn't coded with five-9's of uptime in mind. Changes to analog trunks require a complete restart, which may not be possible in a busy phone system.

I like Asterisk. I think that in the right circumstances, it's a great tool, but you have to go into it with your eyes open. If you're time is valuable, go for a packaged solution.

As far as VoIP, you need to consider two cases:

1: VoIP Handsets on the same network as the phone system. (At least 10Mb/s of bandwidth available)
2: VoIP for inbound & outbound Telco.

My experience has been that VoIP on the local network has worked fine. My phone is on the same VLAN as our production network, and it has all the standard services running over it for ~30 PC's. I have NEVER had an audible artifact related to network traffic, including when I was trying to saturate the link with 80Mb/s of traffic. We're running G.729 for all of our SIP phones.

My experience with VoIP over the Internet has been hit and miss. As long as you have enough bandwidth between you and the VoIP provider, you can expect at least cell phone quality. The problem is if you have any bandwidth constraints or packet loss, you will degrade rapidly. Someone else mentioned the difference between GSM, G.711, & G.729. G.729 does seem to be the best option for us.

there's a reason why things are the way they are (2, Interesting)

Anonymous Coward | more than 8 years ago | (#14144078)

We were in excatly the same boat as you. We considered MANY different options, including what used to be called Centrex, and the IP version of the same thing from two vendors, a SIP client system including Asterisk and some (not all) IP phones, and different IP key type systems.

Of course, it got worse, not better. After a DISASTROUS trial with Cisco, we realized we should have gone with a telephony product vendor ... gee, the SAME one that supplied the OLD key system!

I couldn't wait to get that P.O.S. Cisco thing off my desk. It regularly lost calls put on hold, had display problems, did not work with the power inserter, and often simply DID NOT RING!!! It was worse than my old junk cell phone.

Finally, the Nortel integrator did a weekend overhaul and installed some kind of PBX replacement unit in the server room. I like the phone better, too, MUCH better quality speakerphone than I have ever had before. You can HEAR people!

From my perspective, we should never have considered these other wacky ideas. Having relaible phone service is just too much of a background necessity in business to be playing around with "baubles" and the Nortel people we spoke with seemed to just 'know' the phone lingo and had eveything working perfectly. That was almost a year ago, and it's amazing, NO service calls! Compare that to the 3 times a week calls before.

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