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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
looks like its voip night
Asterisk.
"Think about how stupid the average person is, then realize that half of 'em are stupider than that!" - George Carlin.
... 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).
Fight the fall of slashdot by supporting PlayfullyClever in your sig.
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
asterisk . try asterisk@home for a quick install/demo of asterisk's power
http://www.asterisk.org/ - Asterisk is a good place to start..
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.
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.
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.
Don't think that a small group of dedicated individuals can't change the world. It's the only thing that ever has.
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.
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.
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".
jred
I'm not a mechanic but I play one in my garage...
Or you might get shut out: http://slashdot.org/article.pl?sid=05/11/29/231822 2&tid=215&tid=187
Quality Hosting e3 Servers
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.
DASH911! /good luck.
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.
For very small offices (up to 32 phones), take a look at TalkSwitch small PBX. Prices start around $700 for the entry-level 4 local extension POTS unit. The 8-extension unit with 4 VoIP ports is $1800.
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...
+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!
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.
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
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.
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.
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
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.
I have implemented many offices with Cisco Call Manager, sorry as this is the only VoIP PBX I have experience with. /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.
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
Hope this helps ~MP
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).
... 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.
so, you've installed a sip proxy, it rewrites the RTP packets so they only go out on
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.
Screw the negative karma, anyone who reads Slashdot knows that 90% of the answers are going to be *.
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.
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 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.
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!
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!
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.
Thats not counting phones, network upgrades, and whatever cards you'll need for your asterisk box to talk to things. So figure 10K.
We use a ShoreTel system works wonders you can have a pots or pri feed it. Real easy to manage also.
We're running a Zultys 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.
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.
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.
I read your presentation. How do you spell "evil"?
I spell it "predictive dialling"
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
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.
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 Here is a writeup from crn.com crn.com Info on the individual products in the systems can be found here Handsets Gateway 8 Port POE Switch 24 Port POE Switch 24 Port Layer 3 Switch Router WAP
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.
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.
Karma: Chameleon (mostly due to the fact that you come and go).
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
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.
I have been researching this very topic all day, and now this is in /.
:) and I will not be around the main office to admin it.
i ng_VoIP_With_Linux/
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
I also found this intereting link, about OpenVPN and astericks with 2 T1s
http://www.softwink.com/papers/Installation_Secur
I would love to put the astericks system in the remote office instead of the BCM 50.
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 prefer the "u" in honour as it seems to be missing these days.
contact Peerio (http://www.peerio.com/ by Popular Telephony.
- 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
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.
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.
... gee, the SAME one that supplied the OLD key system!
Of course, it got worse, not better. After a DISASTROUS trial with Cisco, we realized we should have gone with a telephony product vendor
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.
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
We're using this in our office fairly successfully, great conferencing features and receptionist/desktop user controls. I think they have a larger unit as well. http://www.amazon.com/gp/product/B0009XED4M/104-09 62510-1307931?v=glance&n=172282&n=507846&s=electro nics&v=glance
or http://www.xoasisnetworks.com/
We're using this in our office fairly successfully, great conferencing features and receptionist/desktop user controls. I think they have a larger unit as well.
9 62510-1307931?v=glance&n=172282&n=507846&s=electro nics&v=glance
http://www.amazon.com/gp/product/B0009XED4M/104-0
or
http://www.xoasisnetworks.com/
PCI based cards, don't bother with 'em.
Voicetronix claimed support and never came through.
Asterix - the card worked for a while, then one day...poof! No more support for the analog phones. Warranty has not been honored as I'm using FreeBSD and they want to 'know' its not working on one of the linux forks they use.
Mult-Tech had claimed SIP support when I bought the MVP-210, and, well if by support one means being on the upgrade of the week club...yea it was supported. Eventually, it did work with SIP.
Sipura stuff 'worked outta the box'.
We are a small office previously with 8 lines bellsouth and an old nexpath telephone system(vm-to-email in way back in '98 :)
We evaluated asterisk and in order to do all of the transcoding(voice codec to codec translation) and find ing a good reliable provider was going to take a pretty hefty box and some sserious bandwidth, so we opted instead to get a few sipura spa-2002's doing g729a to some providers and hook them up to our legacy system. Works well so far.
Also have a look at the various sip routers out there. Asterisk is a great solution, but it isn't the only way.
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
I finally got to pull an "Office Space" on my old Merlin PBX, atleast the one non working MLX board, but gratifying none the least. I've got 6 POTS lines and 8 Analog DID lines --> AllenTel AT125-SM -->Modular 8FXO,8FXS Rhino Equipment Channel Bank -->Digium TE110P-->Asterisk@Home on CentOS3.5. My experience has been one of learning, appreciation and collaboration. I originally had a problem with my analog DID lines, but the guys from RHINO, www.rhinoequipment.com , walked me through various dialing plan changes, and custom developed firmware for my chanel bank to test and get the DID lines working properly. Analog DID is an, apparently, old technology, but my technology and they went way beyond the call to help me with my setup. Though I did read a couple of books re * there were times when I wanted to try something else, or something with my setup wasn't quite working. In those times I'd go to the various boards and forums. When I wasn't able to find an answer, I contacted Coalescent Systems. These guys developed the Asterisk Management Portal, www.coalescentsystems.ca , they have a very responsive and knowledgable team and worth every cent. Although I came into my * experience knowing next to nothing of telephony, the resources, community and suppliers I have had experience with have made me believe that nearly anything is possible and given me the courage to dream big. It also feels pretty good to know that if I screw something up really badly, I've got support that isn't too far away. The only thing I would suggest is to have a development system too, so that if you screw something up, you don't bring everyone down to their knees, or make back ups regularly. day
We have ~100 people, the phones get a separate VLAN on the switched 100MB network. Of the six T1s four are internet traffic and two are for the phones. The switches are powered, so the phone only plugs into the ethernet port; they have a builtin hub for the PC, so each user only needs one port.
I don't remember the exact prices, but I think the backend hardware and installation ran about $10-15K (though I could be way off, I wasn't directly involved with this). The phones were about $300 apiece. Oh and I suppose the PoE costs a bit extra (though not that much compared to regular Cisco switches).
Everyone gets two lines (with the same, real, phone number - it's also very easy to give 8-12 lines with a multitude of numbers to the administrative people) and all the features they can think of. The grown-ups especially enjoy clicking a number on their PC and having the phone dial it - yes, it's the small things that give managers that adorable twinkle in their eye. The thing will even email you your voicemail as sound files, though I've yet to think of a use for this.
I'd say this thing has paid for itself many times over. Just the ability to take a phone and plug it into any live port and have it retain it's "identity" is a huge time saver (not to mention the ability to log into any phone and use it as your own - that's just plain cool).
At one point we had to set up a temporary secondary site for a couple dozen people: being able to just set up a point-to-point VPN link and get net access AND phones out of it, with just a router and switches on the other site was a huge win.
We've also moved buildings since and only had to reconfigure the backend, the individual phones just auto-updated (even though we got a different block of external numbers).
After the initial setup and a week to learn the system, there's basically no maintenance. New accounts take a couple of clicks and someone to walk the phone over. We have a total of 1 and 1/2 IT people, I'm the 1/2 and I've never had to go near the phone system in the last two years.
sic transit gloria mundi
There are very important questions to ask yourself before going this route. Are you going to be fully going to VoIP? If so, how much phone downtime can your company afford to take? Power outages, network outages, etc... will affect VoIP when it wouldn't affect a POTS. Are you willing to have no phone service for hours on end in case of a failure? How much business could potentially be lost by going this route? So basically any telephone lines/numbers that are critical to the business should be kept on POTS. VoIP is still in its infancy and too many things can go wrong.
It's better to burn out than to fade away
Our company has ported over our primary phone numbers (including 800 numberes) to Teliax http://www.teliax.com/ recently (we were with Vonage, but had constant dropped calls, and clarity issues). It's a great little VoIP company that offers IAX and SIP termination. They allow unlimited simultaneous calls via one account (along with outbound callerid specification/spoofing), and you are simply charged their $0.02/min rate for both incoming and outgoing calls, plus $5/mo per line. Highly scalable and stable, though a reliable internet connection is a must (we have 18Mbps at our disposal). Obviously, we use Asterisk, but there is no obligation to do so, any SIP-compatible offering should work fine. So far Teliax has been extremely reliable, though they are undoubtibly a very small company, but you also get fast/personalized support from their main engineers, not the teir 1 support idiots companies stick you with [*cough*Vonage*cough*].
... One other recommendation is to retain one POTS/PSTN line for 911 calls, it's doubtful any VoIP company will have that down soon, and you don't want internet connectivity to prevent 911 calls in an office environment... you may already have one for Faxing, since no VoIP provider I've seen yet offers T.38 for faxing, though it appears asterisk may be getting some support soon http://bugs.digium.com/view.php?id=5090 so Teliax will probably have support for that once it's in!
Anyhow, that's just my experience. If you don't already have the bandwidth to spare though, it's probably not the right path for you (though bandwidth saving codecs like g726 [instead of g711 ulaw/alaw] will help conserve bandwidth while keeping call clarity, but possibly adding latency)
Why bother with expensive hardware, when all you have to do is install skype ?
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.
Well done Ginel,
I wish I could have done like you did.
Ravenii
PS I play with Asterisk@home and love it.
Frankly, you have to try the ScopServ Web GUI for Asterisk. It's very easy to use and extremely complete. Routing Manager (NPA-NXX), DUNDi supports, FollowMe, OffSite Notification, and many more features described on www.scopserv.com/en/products.php page.
I have sales guys that work out of their homes. If they can run a soft phone on their laptop and do conference calls on my VoIP PBX, and not use a conference call service, that can almost pay for the cost of a VoIP pbx on a 3 year lease.
My developers' desk phones have dust on them. They already have headsets to use skype to talk to [insert native country here]. Who cares what a VoIP desk phone costs if a huge chunk of my user base does not need or want them.
Blanket statements are bad. VoIP offers a variety of benefits - a good number of which translate into cost savings, but there is no one, great solution for everyone. The important thing is that VoIP has hit critical mass, and you need to assess its place if you are looking at buying / replacing a PBX.
ostiguy
You've not given nearly enough information for a phone system designer to help you. Here are some questions that would normally be asked:
How many voice seats do you have in the network?
Are those seats all in one physical location or are some WAN attached?
How many fax machines are there?
How will you get trunks from the telco? Remain pots or are you busy enough to need a T1/PRI? That usually happens at about a dozen trunks.
I know a company with twenty employees who has gone from Cisco ICS 7750 to a Nortel BCN to an Asterix deploy using the Asterix@home distro. I don't hear them complaining. They try every phone that comes along but they're sticking with Cisco 7940/7960s for their desktops.
I'm running a Cisco 2610XM with some FXS & FXO ports running IOS 12.4.5
If you want to email with someone who does this every day respond to this with a post containing your email and I'll contact you.
I am very easy to get along with, but I don't have time to waste being nice to people who are being stupid. -Theo
"Eh? We have ~100 people, in the three years of using VoIP, we've had exactly one problem: a construction desided that our T1 wasn't important enough to dig around."
And did you directly, or indirectly get your T1 from the phone company? The reliability of VOIP is dependent on the hardware it's run over, and while some of the Internet runs over phone company infrastructure. A lot doesn't. The POTS link however is end-to-end high reliability and it shows.
This is a wee bitty redundant, but the figures might be interesting:
One of my clients recently looked into a PABX/VoIP solution for their two very small offices. They required only 10 IP phones and two gatekeepers.
Samsung's quotation was ~AU$14,000; Nortel's was ~AU$18,000. [AU$1 ~= US$0.70]
These were proprietary systems with weak licensing (Nortel: 32 license minimum for voicemail, etc.), limitations (Samsung: only four calls simultaneously!)
Another mob wanted $8000 for just the IP phones necessary, with ongoing (extortionate) costs for using their ISP, their VoIP provider, and their gatekeeper.
My quoted Asterisk solution will be less than AU$6000 for 2 servers, ISDN/PSTN cards, quality IP phones, no licensing, et cetera. Plus the features on offer are more numerous and 100 times more customisable.
Why would you bother with anything else?
My AU$0.02
Asterisk -- 'nuff said.
The secret to creativity is knowing how to hide your sources. -- Albert Einstein
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
I do not normally bash a company but two of the prior companies I did work for had avaya VOIP to connect offices between Atlanta and Gainesivlle Florida. First we went through a frame relay we had lying around and it worked reasonably well, but when we ditched and went on a fractional t with more bandwidth for a few more phones, the quality dropped tremondously, and Avayas answer was to buy a special router for a grand that solve problem of latency, garbled calls, etc. Even after shunting the traffic to a full t, approx 20 extensionions, maybe 8 in use full time, still was fairly crappy.
our developers kludged an ap on a linux box for the call routhing and the problems disappeared.
Avaya sux
Puto
The Revolution Will Not Be Televised
Steadily we contribute to a massive knowledge base revising RFC's, protocols, interoperability standards, and marketability premise while walking the big line under fire from bigger iron and governmental agencies, threats of greasy palmed regulation, and the balance of overall OSS zen. With over fourty config files and an entire platform depenedent scalability, having someone come to your place and show you how its done is worth the money especially in this uncertain interim. I hate to sound like that but I'm in the fire day in and day out and I can tell you from experience that we put together some of the most heavily customized communication systems in Texas and are fast growing enough to realize that we do have a product to sell that is worth the cost of replacing million dollar PBX's, and then some. I was all for asterisk falling onto my desk where I used to work because I was going to be given the task of managing the CLI, but now that I have real time under my belt I don't know how the system administration could have made daylight for it without re-allocating allready precious and specialized resources, even in a multi-million dollar Open Source based operation. Seriously, my comment might wreak of FUD but geez man I'm allready preaching to the choir about it. To make a long story short, unless you got some guys that can really handle it just call us. Yesterday I had to tell some guy that didn't speak English that his Windows 2000 DHCP server went down and it wasn't because of Asterisk, the phones were fine, his XP desktops were just knocked offline, and I'm talking about a guy with a distribution warehouse full of Cisco refurbs.
You are about to give someone a piece of your mind, something which you can ill afford...
And were are the ACD capabilities for Asterisk?
http://www.voip-info.org/wiki-Asterisk+Wishlist
Plus if one's running a call-center one needs to be able to keep track of overall phone system stats, as well as per-agent.
David Mandelstam of Sangoma will give a talk at SCALE 4x about building low cost open-source VoIP solutions.
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 side note to all this is the ease of moving phones and computers around your building.
As a net admin in the UK, I serviced a building with about 150 computers, 80 phones, and 300 jacks. Instead of having hard-wired telco jumper blocks, the telco lines ran to a patch board in the bottom of the same rack with the patch board to the network switches. If someone needed their phone/computer moved, you simply moved the jumper from the old jack to the new one, phone or computer as needed. This worked great with the PBX phones, too. If you can do this, it can save you a lot of telco phone movement headaches and fees.
Better yet, the RJ-45 wiring for 568-A/B have the pairs as pass through, so the same jumper cables worked for phone and computer! At the user end, the RJ-11/14 jacks fit the RJ-45 plugs, no problem.
More often than not, I'd have the jumpers re-routed before the customer had finished moving their stuff down the hall. The only problem I ever had was an unmarked crossover cable I grabbed from the pile for a new installation.
Pacifist paratroopers yell, "Ghandi!" when they jump.
There is some really good basic info from the FCC here: http://get.sent.to/voip
http://tinyurl.com/4ny52
I have spent the past month watching a guy with 30 years of PBX engineering experience try to configure a 12 extension Avaya PBX.
No thanks.
We replaced our Avaya G3 with a Zultys MX250 just recently, keeping our T1 PRI lines for call quality reasons. We went with VOIP for the feature set and price, not for long-distance savings. I'd think the Zultys setup would work really well for a small software company.
I found out about Zultys at commweb.com - they have articles on a lot of alternatives to the big guys (Cisco, Avaya) and Asterisk.
Just a couple of tips from someone who's been there:
- it's a lot harder than you'd think to get a fax machine working on a VOIP system, reliably, so consider electronic fax seriously (unless you honestly don't need fax)
- it's worth your while to catalog all the features you're used to having in your circuit-switched PBX, and ask your candidate new provider how they're implemented, exactly - even the small stuff
The Asterisk site is a great resource, even if you go with somebody else's SIP system. We used Cisco phones but I don't know that I'd do that again, as they've been dropping like flies (20% of the original set have been returned for replacement, and some of the replacements have, too).
VOIP is, without a doubt, a lot more fun to work with than the old circuit-switched stuff.
All the comments I read in this thread are dead on. I'm going to add some things that weren't suggested and elaborate on some others:
1) The best thing you can do to ensure call quality is to start with good phones. I've used a bunch of them and so far I like the Polycom's the best. The SoundPoint IP 301 (2-line) and 601's (6-line) are great quality with a nice price point. I'm sure the Cisco's are great too. Sipura's are a nice price point as well and have some good features. Use ATA's only where absolutely necessary.
2) Remember that modems and fax lines might be a bit of a hassle.
3) If you don't need Power over Ethernet you can save some $$$. The side effect is no phones in a power outage and more cords to deal with.
4) If your internal infrastructure doesn't have enough data cables you'll want to get phones with integrated switches.
5) I'd suggest going with a SIP based system. Cisco's MGCP seems to be falling out of favor and you have a lot more choices with SIP.
6) Having said that, I love Asterisk. It works great. Asterisk @ Home is the best way to get up and running.
7) Is call accounting a high priority? If so, you may need to carefully evaluate that before proceeding.
8) Are you connecting multiple sites? IP trunks are the way to go.
9) Provision your inbound/outbound via normal PSTN trunks. Either single-line (1FB's) pots lines or a T1. If you want inbound caller ID and other good stuff, I'd go with PRI's. Or, set up tie trunks to your existing system.
10) Don't rule out a traditional TDM solution. Some of the basic switches on the market these days are incredibly cheap. Mitel's are one such example.
11) Outline the features you need in a phone/system before installing it. Some things, such as shared (bridged) line appearances are incredibly difficult in VOIP systems.
----- obSig
I initially used asterisks and it worked well. However it become a burden to constantly manage a phone system, especially since I had other responsibilities. The best lesson I learned is spend a little money and save a crap load of time. Basically, what linksys did for the router switchvox will do for the pbx.
If your business is tech-oriented that is to say you have some geeks and at least one uber-geek. And your uber-geek is not already generating massive profit in some other way- ie: s/he has time on their hands then go with an Asterisk system. Your uber-geek would get a chance to have fun and learn and you would get a cheap (not counting the uber-geek's time) solution. Everybody wins.
Otherwise, go with a Talkswitch system.
I can definitely recommend the Talkswitch small office PBX system as a link in your overall VOIP system. I have deployed 2 of them and they are networkable up to 4 units with a total of 32 onsite extensions and 16 lines. The baseline unit is $1500 for 4 lines, 8 extensions.
http://talkswitch.com/
The Talkswitch folks have definitely put some considerable thought into the small business market and what it needs - cheapish (at "business pricing") solutions that are easy to administer and expandable...right up until the point when your business is probably getting big enough to be looking to drop a lot more money on a phone system without any problems.
They take analog incoming lines (currently) so you need to get your VOIP lines with "Analog Telephone Adaptors"... which is how most VOIP vendor sell their product anyway and the cost for the ATA is trivial. Then you plug in any analog telephone saving you the major premium of paying for VOIP handsets.
The administration for these units is done through a simple GUI and is trivial.
We currently using a mix of regular phone company phone lines and some VOIP lines.
Why not get an Asterisk turnkey system?
First off let me say I'm a huge supporter of open source and the business that I work for uses it everywhere that we can. So that is where my biases are.
From what I can see, the current offerings are more expensive than the talkswitch and while they over scads of extra features, I would argue that almost all of those features are utterly inconsequential to a small business. AND the turnkey asterisks do not have one feature that is important - plug and play incremental expansion. Meaning that you have to dump your eg: 16 extension Yonder system when you need 17 extensions (note: I haven't researched this totally).
The Asterisk turnkey systems I am seeing even with their easy to manage GUIs still seem a little edgy and hackerish and *most* small business don't have their own resident uber-geek to solve problems. The day *will* come when some Asterisk vendor sells the perfect replacement for the Talkswitch systems - one that will even supercede it by say offering the ability to grow incrementally to a much larger size, perhaps starting even smaller.. and at a cheaper price. But it is not here yet.
As for the actual question - what provider?
As for the actual question - I am really not sure it matters much. We are using Vonage. The big vendors are well known and you probably know them already.. AT&T, Packet8 etc. Stay away from the little vendors. Don't even go with a big vendor unless they offer a "business" service. Why? It is *NOT* worth the pocket change (for a business)/month that you will save by going with some unknown. It is not even worth the $10/month you would save by trying to use "residential" for your business service (unless of course your business is _really_ small as in being run out of your house with you and maybe one other employee). You are already pushing the envelope by using VOIP. In general it is worth the 10-30% overhead as a business to buy products and services designed for and marketed to businesses and to go with major brands. That doesn't mean you can't shop around, buy on sale etc, but if your business is so strapped you are seriously considering going with
Rudolph Funkmeyer's $6/month VOIP service you should really be wondering if your business plan is working out and maybe you should just fold your hand and give up.
The market seems to be settling around $50/month for the first business VOIP line and $15 after that. With this approach you will save money over POTS for any kind of long distance usage or any more than 2 lines or both.
I have used both Asterisk and sipXpbx http://www.sipfoundry.org/sipXpbx from SIPfoundry. If you are going to go straigt SIP based I would recommend sipX with Polycom phones. A good source of info for sipX is the wiki, found here http://sipx-wiki.calivia.com/index.php/Main_Page.
We are a small US-based company with an offshores office in Vietnam. We use the Packet 8 VoIP service in the US, and I have learned that countries that have protective governments like Vietnam may disallow connections to VoIP IP addresses. Certain foreign ISPs might block all VoIP services. There are still some privately-owned ISPs that allow the traffic through, but who knows for how long they'll allow it. I'm not certain on the laws, but it probably is illegal to even use.
I'm currently setting up a VoIP system for a very small office (5-10 people). What we've got is an Asterisk PBX setup by a company called Fonality. They did a pretty good job doing the initial setup. They will set up everything depending on your outbound config (T1 or whatever) and even set up phones. They can do remote support which I've found tends to be very quick.
They also have a web-based front end for configuration of simple tasks, (e.g. extensions, call menu, etc..), though I don't use it and prefer to edit the asterisk config myself (their config is broken out into lots of small includes, which makes it a bit harder at first to understand the dialplan flow).
The Cisco 7960 phones are great if you have the budget for them. I believe they run around $400 each at the moment. I haven't tried any of the "softphone" solutions yet to see if they are any good.
I would definitely go with a real VoIP provider and not try and use the Voice-Over-My-Internet-Connection route. QoS is a huge deal. We are using MCI at the moment, and things seem to be working out decently well. One thing to watch out for is that if you want to save money and split a T1 (1/2 voice, 1/2 data), I've found that MCI (and maybe others) do not offer Caller ID on the voice side since it is not a full PRI line (we are moving to a full PRI line very soon). Also, getting a split T1 means that you need an external TSU which is around an additional $1000 up-front hardware cost.
The problem with VOIP is where it is you're talking about it. Heck, right now when I call a vonage phone from my phone it's sending voice over ip for part of the connection, anyway. So depending on how far you want to go with your VOIP implementation you might want to check out ENUM and peering with other people to reduce connection costs. You could just do voip in house and go out through the pstn (even not scrapping your current phone system but adding capacity with voip, and using a gateway if you decide to use voip for your service). Where the end ends in end to end is all up to you.
I am in the strange position of having cable broadband, but no landline for the next couple of months, and was wondering if anyone had a cheap solution for me given the hardware I have available.
My father's house has both cable broadband, and a spare landline, which gave me the idea of trying hack together a setup which would let me make use of his landline remotely over the net, using VoIP assumedly. ie. I can place calls remotely from my computer (preferably using a real handset), and any time the remote line rang, it would ring at my computer (or, the above mentioned handset).
I have countless voice modems available to me, but I haven't heard of anyone getting the models that I have to work (supposedly most voice modems are half-duplex, and would only work like a walkie-talkie at best). However, I also have have access to some ISDN equipment, which seems like it would be ideal. I have a 3com TA, and a 3com ISDN LAN "Modem".
Anyone out there with an idea? I've always wanted to hack together a simple pbx, but haven't had much luck in finding leads on google. I'd like to do this on the cheap (ie. not having to buy a proper ATA or FXO) since I have a phone at work and can live without one at home for a few months.
My company has switch to Asterisk PBX (using Digium hardware for the hardware telefony) on a dedicated Dell Optiplex 520GX. We're using it on a small call center of about 30 people.
On this hardware you can only put ONE digium card even if there is 2 PCI slots because of IRQ issues. That's the only problem we had.
On the software side we've started by installing Asterisk at Home (insert a CD-ROM and Asteriks is up and running on Linux) and then we've started to make changes until we created our own dialplan from scratch.
For a budget of $6000 (PC + digium carsds + 22 IP phones + 6 analog to VoIP converters) we have created a PBX that in the market would cost between $25000 (or more) to get the functionality we wanted.
And you know what? It works well for us!
Can't these people form their own opinions? What is the purpose of ask slashdot? Is it solely to coddle the administrators who cannot fulfill the basic requirements of their position by providing them with a forum to mitigate their ineptitude? These administrators never post what they've already looked at, read about, or even that they've looked at the most common solutions via outside providers. They don't even tell you the basics of their company. How big is the organization? How many phone lines do you have? How many do you need? How many dedicated fax lines? What is your current bandwidth usage? What kind of service level agreement do you have? Would the increased usage of bandwidth for VOIP be more or less expensive than another hardware solution PBX or computer system. How many dedicated fax lines? How many POTS lines would you keep in service for failover? Would you actually save money by having inhouse system monkeys working on it? Perhaps you'll be able to reduce IT staffing to make this even more viable and cost effective for the company.
If you're wondering why I am posting as an AC, it is because I do not have an account on slashdot anymore.
We switched from a smallish Bizfon POTS system to an AllWorx VoIP system and have been very happy - both with their server, their software (platform) and their feature 9112 phones. We're currently using 6 CO lines for all outbound communications but have experimented with a SIP gateway via Vonage. We also have four staff located around the country all using the same VoIP phones and connecting right into our phone system. I dial 214 and I reach a colleague in San Franciso; 217 and another in Dallas, etc. Undoubtedly the same things you'd get with other VoIP solutions - commercial or open source. The Allworx box is around $5K and the phones a little over $300 each. While we could have explored Asterisk, the Allworx solution was basically usable within an hour of plugging in. Might have been shorter if we read the manual in advance of playing around. But what are the chances of that really happening. :)
Perhaps the submitter or the editor could do some work and tell us the needs of the company. How many people? How many phone lines in use now? How many more do you need? How many more do you think you'll need with expected growth? What the hell have you looked at already? have you tried google? What about the VOIP wiki?
a slut did tulsa
VOIP equals poor quality. The bandwidth for G.711 is 80K per call versus a normally switched call over a T1 at 64K. T1's cost the same no matter what carrier. I've installed both Nortel PBX's and Cisco Call Managers. The Cisco has poor uptime, poor voice quailty that cracks and pops, and costs a heck of alot more than Nortel or Avaya PBX's. I feel sorry for all you losers out there that have no idea what in the hell you're even doing being involved in voice services for your company. Stay away from Voip, because if you don't. You'll get caught with your pants down being asked why the quaulity sucks. Unless, you and your company is already used to cell phone quality calls. Keep voiping away.
We have an asterisk system driving PSTN on a PRI at my office. We're gradually getting to the point where we're gonna need to decide to either get a new channel bank and futz with the punchdowns or migrate to VOIP.
There are 2 big issues for us:
1) Cost. right now we're supporting 18 people for around $400 US a month plus long distance. We have clients in New York and Germany. For the features we want... that's turning out to be ~$25 a month per head. Most VOIP providers want $60/month for what we have. No Dice.
2) Regulations. the 911 deadlines are one thing. We can deal with that. We're not the only ones in that boat with Asterisk, no doubt the community will straighten that out toghether. CALEA is another concearn altogether. If we bridge our internal pstn architecture and an external VOIP architecture then with current bills circulating in congress, we may have to redo all our stuff at great expense to provide the feds with a backdoor. That simply will not happen. Our clients are multinational technology heavyweights. We will not pay money out of pocket to give anybody a backdoor.
So: In the future I see our company linking offices together with VOIP over some sort of VPN and using voip handsets at the desks. Maybe some logic so if you're calling a different country where we have an office the call routes over the vpn to there first so it becomes a local call. But not now. The regulatory climate in the US is such that if you invest in this stuff now you'll probably get burned (Which is probably exactly what the ILEC's lobbied for) and you may violate any number of NDA's. And right now pretty much all the business VOIP providers aren't cost effective for an SME. For an SME, them's gouging rates.
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
I think you meant s$$$load.
Heh. Me too. It really floored me once I went and downloaded it. I had that thing up and running with a broadvoice VOIP account in like 2 hours. Andrew (the head of the project) has really outdone himself with that project. It just grew and grew and grew. I'm amazed that Digium hasn't caught on to it and and whipped up their own ISO. Ever check the download hits at sourceforge?! Holey cow. Thats a lot of downloads!!! I'm still waiting for A@H to move into the top 10 sourceforge project. I believe its at 14 right now.
- Ginel
Here are the issues we had with the phones:
I wasn't involved with billing, but I got the impression that Packet8 sometimes overcharged for certain things. I can't remember -- not my area.
IronChefMorimoto
I've installed a 3com NBX SS3 system in my wife's law office (~20 handsets) with little problem. We connect to a T1 line which is split 50/50 for voice and data. The 3com system does pretty much everything and is super easy to administer. It's also very reliable and we haven't had to reboot the machine in 9 months of continuous operation. A good selection of hardware is available on eBay but you can also buy it all new from 3com re-sellers. I used eBay and then contracted with mtmnet.com to do the initial configuration of the device. This can be a little tricky if you've never done it before. They did a great job and we haven't had any problems at all since we went live.
> from POTS to VoIP.
n dex.php?id=11.
I have been managing an Asterisk installation at my
company for several months now. The Asterisk PBX has
been rock solid and absolutely amazing. It works so well,
I working on another Asterisk install for a spin-off
corporation as well.
First, background. My father is an old-school
telecommunications manager who frowns upon VOIP. I had
five years in the voice-on-demand (audiotext, IVR)
industry before doing more general system admin and
database work for the last ten years.
Everything you need to know is in O'Reilly's 'Asterisk:
The Future of Telephony'...
http://www.asteriskdocs.org/modules/tinycontent/i
That is a great primer on both VOIP and telecommunications
as well as a strong installation guide for Asterisk. Download
the PDF version and read it before you make any decisions.
Our implementation is a hybrid. While our phones
are SIP (Cisco 7960G) and our PBX is Asterisk, most of our
traffic is carried on a PRI. Local and long distance calls
run across the PRI. This gives us very reliable service and
good voice quality. Plus, a PRI (with tens of thousands of
minutes a month of long distance included) costs about the
same (or less) as the bandwidth necessary to support the
VOIP calls and VOIP-to-telco provider.
For our international calls, we do have accounts with
a few VOIP-to-telco providers and route those calls over IAX.
I wouldn't go entirely VOIP if phone calls are important
to your company. As often as one in seven tries, our VOIP
routes fail for one reason or another and rotate to the next
provide. For the few international calls as we do, our users
rarely notice. If we were using VOIP for all our calls, I can
see these spurious anomalies as being a huge problem.
The advantage to Asterisk as a PBX is not so much its
ability to provide dialtone at a reasonable price. Even a
commercial PBX can do that at about the same price point.
The advantage to Asterisk is that the extras are free.
Voicemail isn't an added cost. IVR isn't an added cost.
Having Asterisk pull its caller-id data from your CRM
solution (in our case, SalesLogix) instead of just using
the telco-provided data isn't an added cost.
My father still swears by Ma'Bell. And in terms of
absolute reliability, he's right. Ma'Bell can get you five
nines year after year. A well-configured, well-administrated
Asterisk system with PRIs (instead of pure VOIP) is close but
still isn't quite there yet. But, by the time you add in all
the additional costs for a commercial PBX, Asterisk is by far
the less expensive solution.
I'll take four nines in exchange for tens of thousands
of dollars savings a year.
Matt
I have a client who does litigation consulting. They have two offices (one is considered a remote) in two different cities, and a Toshiba Strata CTX phone system that has a VoIP card in it. They have four remote users at the remote office using one keyset and three soft phones, all using the G.729a codec. Both offices have T1 connections. The client wanted VoIP traffic to occur over a secure link, so I'm routing the traffic over their existing VPN, which is run between two Cisco 506E Firewall/VPN appliances.
I tried to explain to them that it would be easier for a snoop to hook up a butt-set outside their office (they use POTS for phone service) than it would to find and capture their voice packets, but they're insisting on a secure method. They are experiencing some call quality issues at the remote office, and I'm concerned that it may be the VPN.
Can you offer any insight on the matter? Much appreciated, sounds like you've been through the mill.
There are a few (though not many high quality ones out there yet) outsourced ip phone system providers for small-to-mid-size businesses out there. One of the highest rated and ahead-of-the-game companies out there is M5 Networks that currently primarily serves NYC area. Data/voice supplied and managed, superb customer service, redundancy, and no need to purchase/manage/bother with phone/long distance/ISP companies, and still manage your own network and do what you do best. The voice service is specialized, versatile, feature-rich, and it's what companies like M5 does, and does best.
Give hosted companies like this a look while you're researching - they're well worth it.
First and most important, is quality of calls. If you want to ensure voice quality, it is best to seperate your voice and data networks. They should be seperate lans, or at least VLANs if your switch can handle that and ensure that something like SQL Slammer won't fill the backplane on the switch and impact voice as well as data.
Second if it is ever necesarry for the voice traffic to be "trunked" over the same link as the data at any point. You will want to ensure you put some sort of traffic shaping box in there to reserve as much bandwidth as you need for your voice traffic. (about 80 kbps per concurrent call for uncompressed voice) If you use GSM, or any of the other different companders/compression you can change that number, but it will change the quality of the voice.
Phones: I like the Polycom SIP phones, the Soundpoint IP 501, is a 3 line phone, with a great speaker phone, and is a reasonable cost compared to some other phones.
If you are looking at Asterisk, study voip-info that is where all the info is. Also know that until you get everything working the way you want it to work, at least someone in the IT department will be kept busy. People need phones to work, and they have certain things they expect out of them.
example: If you don't put a 1 or 2 second pause of dead-air into the line between when a user makes a selection on the AutoAttendent and when the next AA or VM starts talking you will miss the first fraction of a second of the announcement because it will start before the user takes his finger off the button. And there will be tons of other little gotchas.
There are plenty of companies out there that will let you outsource your PBX and have IP phones on your premesis, and some even make financial sense (mainly because you don't have capital expenditure, and need to learn anything except how to run the handset). This doesn't sound like what you want, but it is worth checking them out. (full disclosure I work at CoreDial so I of course wouldn't mind if you checked them out.)
Logical Disconnect
I know its proprietary but I have to say, we just had an Inter-Tel system installed in our offices and it works great. Our main office (read: old) is still using a digital system but the other two offices are connected to our main office via VOIP. In our other two offices there is simply a 1U system (Axxess 5000) and a few POE switch and thats our entire phone system. The phones work great and we have an couple IPRC (Internet Protocol Resoure Cards) in our system so people can take IP phones home (or wherever) and make calls as if they were in the office.
Our current configuration has PRIs at each location with inter-office calls done via VOIP.
Especially once you combine this system with their Unified Communicator (call routing w/ greetings etc based on caller id, custom greetings, collaboration, find me follow me, presence management, desktop sharing..etc) The thing I love about that is that if joe blow calls in I can have my desk phone ring, and if i dont answer, it goes to voicemail. If my boss calls in and I don't answer then system picks up the line and says "Hi XYZ, hold on the line while the system finds me" and then attempts to call my cell phones or other locations and then will finally dump to vmail if It still cant find me.
The other thing I like about Inter-Tel is the fact that they can do their own financing. In our case we are leasing the equipment which makes it a much smaller amount of money each month and that cost covers service calls, insurance on the hardware, loss/theft, software upgrades, etc. So in 3 years we can look at what technologies we are using/not using and change our configuation from there.
Our business size is fairly small - about 200 people across 3 offices (10+ miles from eachother).
Again, while proprietary - its a total solution that, if leased, can be incredible cheap to operate (on a monthly basis).
-
aphex
I Steal Music!
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.
At the company I work for, we switched over to an IP Office system a few months ago. We still use PRI for our connection to the outside, but internally, it's all digital. (Actually, we're using their digital phones, not the IP phones. You might find that you like that better. When I connect remotely to the network, I just run a software phone app through my PC. It's OK, but sounds like a cell-phone when you talk. Also, don't have the mic levels up too high, or you get a lot of echo with most crappy sound cards.)
The admin interface for Avaya's system is easy, so managing it in-house should not be a problem.
Before you move to IP for the signal, though, think long and hard about the kind of network you have. Can it handle the voice traffic without affecting other services ont he network? Will those services affect your voice traffic? Are your switches capable of managing this for you? In our case, we already had separate CAT-5 cabling for voice since our old system was digital. This makes traffic management practically a non-issue.
As others have mentioned, Asterisk could be a good solution, but be sure you know what you're doing and have time to learn the system.
Be quick to listen, slow to speak, and slow to anger.
http://www.kx-td.com/
We used them about 6 months ago for a VoIP phone system solution. We have 25 employees, and 30 telephones, plus voicemail. Couldn't be happier.
- Doug
I have been supporting 3Com voip systems for about 4 years now. Stay away. Stay away.
Of course as many have pointed out "Asterisk" is the answer. The poblem is someone who runs, say a shoe store will not be able to set up an Asterisk server himself and will need to hire a consultent. But this is not different from seting up a point of sale cash register application or heck, even painting the office interior. For those who don't like Asterisk there is a new "fork" called "Open PBX" that usesAsisk cose but making it truely open and "pure GPL'd" While this gets you removed from Digium's comercialium it makes seting up a PBX even more of a do it yourself project. Open sourse does not mean "free as in free beer" you will need either your time to learn and set it up or you pay someone else. The good news is that nerds with free time really can have a hugely complex phone system with all the bells and wistles for just the time it takes them to implement it. Even for a single guy living in a small apertment Asterisk is usfull. It's nice to get voice messages emailed to me. I hate going through the answering machine's messages one at a time. It is worth instaling Asterisk just so I can see the messages inside a web browser and click on the ones I want to hear without need to leasten to them all. Also access to the messages from any phone or PC world wide is worth it too., I hate those little tape machines. THere are other questions to be answered even after you deside to use Asterix/Open PBX. For exampleyou will need to select one or more VOIP service providers, these are companies that coonect VOIP to the PSTN
http://help.fonality.com/?id=171025
If you don't already have native support for VOIP in your PBX, you might be able to replace your whole PBX system with one like PBXtra from Fonality for the same price as retrofitting your legacy PBX.
http://www.fonality.com/
Some other high level considerations:
I'm pretty much the only tech in the office, and I had not time to hit the Asterisk learning curve. Could not find a single asterisk installer in the UK. Not one.
Anyway, all the VoIP stuff I saw was shite, or expensive. The one we chose, the zultys mx250 is not too shabby, but it isn't that cheap, although their phones are reasonable, and don't look like sci-fi monsters. It is linux (on PPC) under the hood, and has reasonable levels of UK support.
Tom Newton
Every time I look into this I pass (but I look every time I have to revisit it; one day VOIP will be ready). I mean, you can make it work, and make it work well, but is it core to your company's mission? Phone service itself probably is core (it is to most, but not all companies), but is having people grok everything about the phone service a core part of your company's business?
What I've ended up doing every time so far is just buying a used PBX. They get cheaper all the time. They aren't always all-singing-all-dancing, but actually that's usually a good thing. Somebody still will need to learn a lot about the phone system, but less than they would have to learn were they building a VOIP infrastructure.
Generally I've been much happier with ones that don't need proprietary handsets.
Oh, and personally I've been screwed too many times by Altigen.
You may find a product by a company in Ottawa, Canada quite useful. The company is Nimcat Networks (http://www.nimcatnetworks.com/ and they produce a small/medium business solution that is complete Peer-to-Peer SIP. They produce the software for embedding into IP Telephones - presently their software is available from Aastra (http://www.aastra.com/enterpriseip/pro_228.asp), and apparently it will be available from Avaya IP phones in the near future (actually - it seems Avaya may have acquired them). Well ahead of the Linksys One announcement recently (as it doesn't require a centralised Cisco box) and also supports a PSTN analog device (4 lines) for incoming and outgoing lines.
Have you or anyone else heard anything about the SBC hosted VoIP offering called HIPCS?
They apparently have a few installs in teh Chicago area, but not finished isntalls here in Central Illinois. When they presented it it sounded very much like the Speakeasy solution with them hosting it and we going with the same Edgemarc firewall and Cisco router, switches and phones.
I am definately interested in peoples reaction to there service both good and bad.
I'll never forgive you for turning your magazine from a great resource for Mac developers into yet another consumer Mac mag. You shouldn't be reviewing VoIP: you should be writing about how to develop VoIP apps or something.
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.
"Nortel leads the recovering North American IP PBX market," said Matthias Machowinski, directing analyst at Infonetics Research, in a statement. "Avaya and Cisco are engaged in an epic battle for second place, with Cisco steadily encroaching on Avaya's leadership position." Nortel offers a SMB IP PBX - Business Communications Manager - for a very reasonable price (in general, Nortel offers more competitive pricing than Cisco) and Nortel has a very strong R&D culture and tremendous products. By purchasing a SMB IP PBX, you can do your trunking in-house. Having an off-site third party company manage the connection to the public telephone network is risky...