Here’s the scenario…….
You’re planning planning ahead for a consolidation of your business phone systems including a potential move of your headquarters to a new building.
Currently your company has 300 employees and operates in 15 locations:
– 6 warehouse locations with business offices
(~30 – 50 employees each)
– 1 small warehouse (5 employees)
– 2 business offices weebo (~10 employees each)
– 7 small stores (3-4 employees each) – 2 share space with warehouse locations
You also have some outside sale folks that work from home most of the time.
Currently you run several disconnected phone systems and some Centrex (store locations). You’d like to standardize on one platform with integrated voicemail for the company. The plan is to do this in the next 1-2 years, whether or not you move to a new building.
All of these facilities are connected data-wise via a private routed network served by a Tier 1 carrier. Your headquarters is the hub for these locations and currently hosts all of the data servers.
When and if you move to a new facility your boss is considering outsourcing the mainframe and server systems such that all of the equipment is hosted by a separate company. This would relieve you of the considerations of building a server room in the new place. You do currently have a raised-floor server room, where your current phone system is located.
Of course with no server room (if you go that route), this limits your ability to host a PBX (you currently use a ROLM 9751).
Here’s the questions you should like ask….and ensure answers for:
1. In a hosted PBX or VoIP solution, livewebdir or even with a centralized on-site PBX can you still keep local numbers for each location?
2. If your equipment is centrally located, how do local calls work? e.g. – if your phone system is located in Maryland and someone in New Jersey needs to make a local call, is that really a long-distance call since the equipment is in Maryland? How is this typically handled?
3. What about DID numbers? Can you keep these? How are they routed?
4. What would a company do in terms of having a local operator at larger locations? Is there a sort of gatekeeper in place at these locations, or would it all be centralized at one site?
5. Currently you use a different automated attendant setup at a few of your locations. Would this still be possible or even recommended?
6. What is the usual way of connecting multiple sites to a centralized telephone system? What type of backup links are typically used?
7. You figure moving to a completely new system would cost around $1,000 per user (phone equipment, huntingtime initial setup, roidirectory new phones, training). Much less for a hosted system, but a high MRC you suppose. Is this estimate in the ballpark?
8. What recommendations can you expect on what type of systems may “fit the bill”? Some features you’re looking for are below:
– Outside sales would like to be able to forward their lines to a home/cell phone.
– Internet access to change user settings would be nice (web-based user management).
– You have several Inside Sales queues, shayarism so you’d need good ACD capabilities.
– Ability to dial by extension to anyone at another location.
– Distribution lists for voicemail.
– Custom on-hold messages by location (different or store locations).
– Local paging at your warehouse locations (page over intercom).
– Local directions to your supplier truck drivers.
– After-hours/emergency messages need to be customized by location. (For example, if your Pittsburgh office is closed due to snow).
9. What about backup analog lines? Since you have a large inside sales presence, the ability to receive phone calls is critical. What is a good number of lines (percentage of total trunks, perhaps?) that are required and how are they usually setup?
Now there may be a number of choices to select a solution from…but in the interest of simplicity and brevity for this article we’ll focus solely on Asterisk. You can apply others to the questions posed above on your own….if you’re brave enough.
Asterisk an open source (free) soft-PBX type program, that can do just about anything. If you choose a proprietary vendor’s product, some or all of this may not apply, as the following reflects how I’d suggest set up using Asterisk.
I am going to assume your system is all easily routed (no NAT) and at least the server can get on the Internet from your main datacenter. Also, how much bandwidth does this provide you? At full quality (G.711 ulaw codec) a call takes about 80kbit/sec including overhead. When highly compressed (gsm, iLBC, g.729) it can be as low as 10-15kbit/sec.
Plus….stop thinking about this as many small systems and start thinking one big system. Additionally….with IP lines there is often not a channel limit, you are only limited by your bandwidth.
With 300 users, you won’t need THAT much to get on asterisk, in certain situations. A good-size box running Asterisk should be able to handle 300 concurrent calls without too much of a problem. If you do “difficult things” (codec translating, conferencing, etc) this number goes down. The point is you may very well be able to fit much/all of what is required into your existing datacenter. If you require “large things” (channel banks, large PSTN interfaces, etc) this may not apply. Wiki for a page called Asterisk Dimensioning for info about who is using what hardware and what it can handle.
Set up your Asterisk server(s). Standardize on a few models of IP phones (make sure 1 is Uniden…one of the more reputable and capable). Configure DHCP for your network to provide a tftp-server. Probably also set up some kind of database for phone configurations. Use this to make files for the TFTP so the phones will configure themselves. Takes a bit of doing but makes setting the phone itself up VERY VERY VERY easy, just plug it in (assuming its provisioned in the server first).
You plug in the new phone. DHCP provides it with a TFTP server, techquisys from which it fetches a config file based on its manufacturer or model and another based on its MAC address. It also downloads new firmware if your server provides it. It then reboots if needed and the settings take effect.
The two files let you set settings for everybody (what server to use, etc) while defining individual settings (SIP login, softkeys, etc).
You can also distribute things more. Set up small regional or local Asterisk boxes that handle certain areas.
Lastly, keep in mind that how you terminate your calls does not have to be VoIP even if your PBX is.
Now to your questions.
1. You can almost certainly keep all your local numbers, although this depends on what VoIP provider you get. Often one provider can port a number while another can’t. If you have PRIs or something in an area and want to keep them, you can. Asterisk will handle this fine and I’m sure so will other packages. You just need an appropriate interface board.
2. Again, depends on providers. Often you can work out a deal whereby local calls will be free. Even so, VoIP minutes are not expensive, usually 1-2c/min tops and you can negotiate a better rate if you use a lot.
3. See above. DIDs are easy, they will be routed to your central PBX and from there to your sites/phones. If you have local PBX’s they can register directly to the provider if you have different accounts. If you mean real DID numbers (call number, dial extension, get person) that can also be done.
4. You can have an operator on VoIP. IP phones are available with a lot of buttons if you need them (Cisco, Snom 360, and Grandstream 2000 all support sidecar modules). Calls can ring the operator and/or go to wherever you wish based on whatever criteria (time, operator logged in, etc).
5. Sure it is. You can route a call based on what number it came in on, what caller ID was provided, what day/date/time it is, what setting is set to what, or any combination of the above / almost any other criteria you can think of.
6. As above, you can super-centralize or you can spread out with smaller, local servers. Asterisk servers can trunk calls to each other via SIP or IAX2 (inter-Asterisk exchange) protocols. You can route calls based on extension range (2xxx is NYC, 3xxx is Boston, etc) or simply by which server has it (Wiki for DUNDi). All the transport will be across your chosen network provider. Installing backup links is the same as backup Internet links.
7. A bit high but not a bad estimate. Running financials, say $3000 for the main server (assuming you centralize), $300 or less per phone (user), plus man-hours, training, etc. If you need to upgrade your network provider links or switching capacity this goes up.
8. Asterisk can do all of the stuff you mention. Few gotchas…
– call forwarding requires some setup, this can be done by making an Agent for each user or with a call forward script. It is quite possible though.
– web based admin – Asterisk itself can be configured from flat config files or through a MySQL database. There are however packages (asterisk+stuff) that provide a web front end. For example: Trixbox.
– Dial by extension will not be a problem as long as you have the bandwidth to handle all the calls.
– Voicemail distro lists are easy. Make an extension that dials like VoiceMail (mailbox1&mailbox2&mailbox3) etc.
Asterisk supports MOH from mp3 files or other audio files. You can change MOH classes per-channel, per-user, or per anything else. Each MOH class is a folder with file(s) in it. You can make as many classes as you want.
– Paging is also easy. If you have an Asterisk server onsite, hook it’s sound card up to the paging system. Otherwise you can page through phones (most phones support intercom/paging). You can page through an overhead system using either the sound card, or something more specialized – you can get paging controllers with a POTS interface (hook it up to an ATA (analog telephony adapter, ethernet on one side, FXS (station) POTS port on the other), or you can also use a VoIP phone to interface a paging system. Grandstream GXP2000 phones for example have a 3.5mm jack which can be easily wired up to a paging system.