Does Commercial VoIP work for business?

Yes you are correct feptias; my solution does involve replacing the phone system with a dedicated asterisk pbx.

For us this was a simple choice. Two servers (so we have a complete hardware backup) and two 110P PRI interface cards costs about £2000 and we purchased the Grandstream GXP-2000 phones which came to about £1500 for 35 of them. So the £3500 it cost us to install the new PBX was about the same as 3 months lease on our old Lucent Index phone system which had just expired.

We had to put the new system active quite quickly so we went from testing with a few users and calls over voip only to a fully operation system with 35 users and a ISDN30 line in about 48 hours. I was a bit apprehensive about doing this but the switchover went without a hitch.

If you want a really reliable system I would recommend that you go for something like the Cisco phones. The GXP-2000 are good phones but they are not as robust as Cisco.
For example the handsets seem to be not particularly reliable. Out of the 35 we have we have had to return 2 or 3 which have stopped working.
We have a PowerdSine POE adapter which sits infront of the switches and adds POE to the network and we use this so that we dont have to use the separate power supplies with the phones and this also means the PowerdSine can be protected by a UPS so the phones carry on working for a while in the event of a power failure. We have a couple of phones which were running off their power supplies and both of these after a year or so started crashing but they worked fine when they got their power over the network. I don't know if the power supply developed a fault or what but we always provide power over the network to them now.
Having mentioned the minor faults with the GXP-2000 they do have excelent features such as a customisable display, downloadable phonebook from a server, and excelent provisioning.

The above comments are very helpful. As I am not familiar with phone system and VoIP, so my questions may appear to be very naive. I need more help to get my question straight.

Just to add more information about my environment:
Head office:
Internet: E10
10 phone (analog) lines, 20 extensions (analog phone system)
7 branches:
Internet: ADSL with static IP (download/upload: 3Mbps/764Kbps)
Average 4 phone lines (small phone system with rollover) + 1 fax line
Internet usage is not high.
Most branches are in the same city, so long distance cost is very small so far, but it is certainly good to lay a good infrastructure for growth in different cities or even different countries.

Goal: cut communication cost for head office and or branches.

Q#3: What might be a good solution to save money while maintain voice quality and reliability?
Q#4: How should I test it out safely if the solution works or not? I try to avoid running into embarrassing situations when trying to save money for company.
Q#5: Can someone give me more advice and share more of your thoghts/experience?
Thanks a lot.

Perhaps a good starting point would be to analyse your call traffic, maybe using a call logger. If a significant proportion of your phone bill is accounted for by calls to a branch office, then you could try installing a pilot system that just links the head office and that branch over the Internet. This would have good potential for saving money because the VoIP calls would be free (other than equipment setup costs).

The solution I would probably select in this case is an Asterisk server at the head office and another at the branch. I would connect them to the existing PBX at each site through FXO ports (on Asterisk) and analogue extension ports on the local PBX. Callers from the head office wishing to call someone in the branch over VoIP would first dial the local extension that is connected to their local Asterisk server. They would hear dial tone and be able to dial any extension at the branch office (sort of a DISA service). The Asterisks would connect to each other using VoIP (SIP or IAX) over the Internet so the call would be free. You would need help to install and set this up - it is not an out-of-the-box solution.

To add to feptias's reply you would also need to make sure you have some QOS (Quality of Service) at the remote sites otherwise any large downloads may cause the voice to become choppy.
In order to get QOS you need a good router such as a Cisco model however even then you cannot guarantee QOS will get rid of problems as you cannot control the incoming data. Therefore if you intend to rely upon VoIP for calls between offices then you would be best off getting a 2nd ADSL line dedicated for voice.

One more addition to the above, make sure the system you put in place if you decide to go with VoIP has a reliable fax solution. Also account for any other analog devices such as alarms, point of sale devices or modems that you may be using. Some VoIP systems to not handle fax very well. Typically you'd look for something like T.38 support. For those calls you also want to ensure you use the proper codec (in this case G.711 for fax). Your voice calls can try to negotiate to a codec with better compression, such as G.726 or G.729 so that you save bandwidth usage, but be careful about the fax setup.

As grblades suggests a second connection, dedicated for voice and an ISP that would be willing to help you with QoS (some ISP's also do VoIP so they may have that setupon their network) may also be good. If your ISP is the same for all of your offices then you may be able to prioritize voice calls between the branch and main offices better than if the have to travel the public Internet.

Also, along the lines of a voice analysis, look at how many calls you have to and from the branch and main offices at any given time. If you have only a few calls at the branches but more than say a dozen calls at the main office, you may want to consider augmenting the bandwidth there. General rule of thumb for QoS is something like 40% of bandwidth should be prioritized for calling (signalling and media) before you start to see data perfornance degrade and have the possibility of jitter and delay on the calls themselves, so consider the queueing when you look at implementing QoS as well.

Hi All,
After reading all the above comments, I have a better understanding of the topic now.
Currently we press line button (Line 1, Line 2, ...) to select a line when making outbound calls.
Q#6: Is it possible to keep the current PBX and phones while cutting off some traditional phone lines and 'hook up' some VoIP lines at the front trunk side?
Q#7: Is it possible to keep some lines associated with traditional lines, for example, 'Line 1', 'Line 2', while 'Line 3', ... 'Line 9' are associated with VoIP lines. I can ask users to choose 'VoIP' lines when making outbound calls, especially long distance calls, while leaving 'Line1', 'Line2' awaiting inbound calls.
Q#8: Is it possible to rollover from the main business phone number which is associated with traditional lines, to the VoIP lines?
If all the above works, then I might be able to save some money by partially implementing VoIP. Please share your experience. Thanks.

Mixing voip and plain lines is no problem.  There are several ways to do it depending on the final result you need.  It can be as simple as using ATA adapters or voip gateways which connect to the network but provide a plain phone line output.  So this can be connected to a PBX or even just plain old phones.  

A more advanced approach is to install a hybrid PBX which is connected to the network AND the original phone lines.  You then connect your existing phone system to the hybrid unit just as though it were a voip gateway.  This allows more control as now you can switch phones lines around totally transparent to the old PBX.  

By running all phone communcations through the hybrid PBX you can completely control what lines are used for.  So it can be setup to actually decide what line to use based on the number dialed etc. without the user really even knowing it is happening.

In some cases it is also possible to ditch the old PBX but keep the analog phonesets and connect them directly to the new hybrid PBX.

You can call forward you original business number to a voip number.  Actually moving your number to a voip provider may or may not be possible,  depends on the voip provider you use.

If you look at voip seriously you should also take a look at other possible uses of telephoney within your company.  Many features that were not available or were too expensive are now affordable and within reach even for small companies.

Another thing to consider,  what is the current lifespan of your phone equipment?  Voip phone service may or may not work for you today.  Replacing dated phone equipment with hybrid equipement  that is 'future proof' may make sense though.  Then you can swap out POTS lines for voip at your leisure in the future and get updated PBX functionality today.

My PBX is old, but no plan to upgrade yet. That's why I try to keep everything unchanged  except replacing a few phone lines. One traditional line costs $30/month, while VoIP line costs only $10/month. Now here is what I intend to test VoIP line with my PBX. Please let me know if it is a good way or suggest a better way.
1. Apply for a VoIP line. Should I use ATA adapter to hook it up with PBX?
2. How to hook the VoIP line to the PBX? I never work on the PBX. It is always the telecom rep to make the change. Any technical detail/article for such wiring task?
3. Now suppose the following:
Main business phone number (890)1234567
Line 1: (890)1234567
Line 2: (890)1234568
Line 3: (890) 1234569
S#1: When first call comes in, it searches idle line and picks line 1;
S#2: Now second call comes in (while first call is still active), it searches idle line and picks line 2;
S#3: Then third call comes in (while previous two calls are still active), it searches idle line and picks line 3;
This is how it works currently with pure traditinal lines.
Suppose I replace traditional line 3 (890)1234569 with VoIP line (890)1234560.
Is S#1 still the same?
Is S#2 still the same?
In S#3, will the call rollover to VoIP line? Given I get Line 1-3 from telecom#A, VoIP line from telecom#B. If not, what I should do to make such rollover work?
If such rollover to VoIP lines are easy and 'free', then I ultimately keep only (POTS) Line 1, and add as many VoIP lines as needed to save money.
Certainly considering the reliability of VoIP, it may be a good idea to keep more than one POTS lines to maintain enough communication capability in case VoIP lines are down, which is suggested by the above comments.
Please let me know.
Thanks.

It sounds like you have a KSU type system. That means it should be easy to add a VoIP line that can be selected by users as needed. Initially, you could try an ATA/Gateway that has an FXO, an FXS and an ethernet port (e.g. the Sipura SPA-3000). It would require some simple re-wiring to connect it into one of the existing lines - the KSU analogue trunk port connects to the ATA's FXS port; the actual line from the CO connects to the FXO port; the ethernet port must be connected to an IP network such that it has access to the Internet. Inbound calls arriving on the analogue line would still go through to the KSU system as before. Inbound calls from your new VoIP service would appear on the same line on the KSU. When you select that line to make an outbound call it would go through the ATA/Gateway and be routed according to the rules configured in the dialplan of the ATA.

As for the rollover from your conventional lines to a VoIP line, that is something you will have to ask your existing PSTN service provider. I cannot see them being very keen to help though, because you want to do this so you can reduce the number of conventional lines and pay them less every month for line rental. That's not much incentive for them to be helpful, unless of course they are charging you for every call that is overflowed to the VoIP service in which case it may end up costing you more! You may have to work on the basis of keeping sufficient conventional lines to handle all your inbound call traffic and just use the VoIP lines for outbound calls. If the VoIP proves to be so good you think you can justify a complete switch-over later, then that option should be available to you. However, I doubt that you will want to do that for reasons of reliability and voice quality, plus other potential complications.

The call forwarding may be a pain to setup.  It's not so much that the phone company doesn't want to or not,  the reps are often just not knowledgable about features and have trouble finding the product for you.

The exact way it operates can vary from one company/area to another.  In general you are looking for call forward on busy.  This would be attached to the primary incoming number and will dump calls to the alternate number when your whole hunt group is busy.

I recommend experimenting with it to make sure that what you get works as expected.

If you want to just add a few lines or replace lines a few ATA units work well.  It may be less trouble to buy or rent from the voip provider.  This way they will take care of the programming.

Replacing a line just means finding how it is connected now and changing the connection to the ATA adapter from the incoming phone line.  However you may run into a problem if you are adding more lines,  existing equipment may be full or may need to be reprogrammed for the extra lines.

For your locations with only a few lines it is probably really simple as they likely are just using equipment with jacks or plain multiline phones - so just unplug and replug, done.  Your main site is probably more complicated with direct wired punch down blocks, no labels etc. and somebody with knowledge and proper tools would be a good idea.

I will check out the current PBX and find out how it is connected, then I will post it here. Thank you very much for your helpful ideas and experience.

I really appreciate your wonderful tips and experience. I just close it for now. I may post it again when I am ready to try it out. You already provided me excellent answers. Thank you.

I feel each answer is worth more than 500 points. Unfortunately I can only assign 500 in total. Thank you for your help.