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How To Solve A Consolidation Of Your Business Phone System Needs With An Asterisk VoIP Solution 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 including operates in 15 locations: - 6 warehouse locations with business offices (~30 - 50 employees each) - 1 small warehouse (5 employees) - 2 business offices (~10 employees each) - 7 small stores (3-4 employees each) - 2 share space with warehouse locations You additionally have some outside sale folks that work from home most pertaining to the time. Currently you run several disconnected phone systems including some Centrex (store locations). You'd like to standardize on 1 platform with integrated voicemail for the company. The plan is to do the 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 including currently hosts all pertaining to the data servers. When including if you move to a new facility your boss is considering outsourcing the mainframe including server systems such that all pertaining to the equipment is hosted by a separate company. the will relieve you pertaining to 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 absolutely no server room (if you go that route), the 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, or even with a centralized on-site PBX should 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 including someone in New Jersey needs to make a local call, is that actually a long-distance call since the equipment is in Maryland? How is the typically handled? 3. What regarding DID numbers? should you keep these? How are they routed? 4. What will 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 will it all be centralized at 1 site? 5. Currently you use a different automated attendant setup at a few of your locations. will the 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 will cost around $1,000 per user (phone equipment, initial setup, new phones, training). Much less for a hosted system, but a high MRC you suppose. Is the estimate in the ballpark? 8. What recommendations should you expect on what type of systems may fit the bill? Some features you are looking for are below: - Outside sales will like to be able to forward their lines to a home/cell phone. - Internet access to change user settings will be nice (web-based user management). - You have several Inside Sales queues, 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 regarding 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 including how are they usually setup? Now there may be a number of choices to select a solution from...but in the interest of simplicity including brevity for the article we'll focus solely on Asterisk. You should apply others to the questions posed above on your own....if you are brave enough. Asterisk a particular open source (free) soft-PBX type program, that should do just regarding anything. If you choose a proprietary vendor's product, some or all of the 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) including at least the server should obtain on the Internet from your main datacenter. Also, how much bandwidth does the provide you? At full quality (G.711 ulaw codec) a call takes regarding 80kbit/sec including overhead. at the time highly compressed (gsm, iLBC, g.729) it should be as low as 10-15kbit/sec. Plus....stop thinking regarding the as many small systems including start thinking 1 big system. Additionally....with IP lines there is often not a channel limit, you are only limited by your bandwidth. With 300 users, you could not need THAT much to obtain 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) the 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) the may not apply. Wiki for a page called Asterisk Dimensioning for info regarding who is using what hardware including what it should handle. Set up your Asterisk server(s). Standardize on a few models of IP phones (make sure 1 is Uniden...one pertaining to the more reputable including capable). Configure DHCP for your network to provide a tftp-server. Probably additionally set up some kind of database for phone configurations. Use the to make files for the TFTP so the phones could 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, from which it fetches a config file based on its manufacturer or model including another based on its MAC address. It additionally downloads new firmware if your server provides it. It then reboots if needed including the settings take effect. The 2 files let you set settings for everybody (what server to use, etc) while defining individual settings (SIP login, softkeys, etc). You should additionally 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 should almost certainly keep all your local numbers, although the depends on what VoIP provider you get. Often 1 provider should port a number while another can't. If you have PRIs or something in a particular area including need to keep them, you can. Asterisk could handle the fine including I'm sure so could other packages. You just need a particular appropriate interface board. 2. Again, depends on providers. Often you should work out a deal whereby local calls could be free. Even so, VoIP minutes are not expensive, usually 1-2c/min tops including you should negotiate a better rate if you use a lot. 3. See above. DIDs are easy, they could be routed to your central PBX including from there to your sites/phones. If you have local PBX's they should register directly to the provider if you have different accounts. If you mean real DID numbers (call number, dial extension, obtain person) that should additionally be done. 4. You should have a particular operator on VoIP. IP phones are available with a lot of buttons if you need them (Cisco, Snom 360, including Grandstream 2000 all support sidecar modules). Calls should ring the operator and/or go to wherever you wish based on whatever criteria (time, operator logged in, etc). 5. Sure it is. You should 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 pertaining to the above / almost any other criteria you should think of. 6. As above, you should super-centralize or you should spread out with smaller, local servers. Asterisk servers should trunk calls to each other via SIP or IAX2 (inter-Asterisk exchange) protocols. You should 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 could 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 the goes up. 8. Asterisk should do all pertaining to the stuff you mention. Few gotchas... - call forwarding requires some setup, the should be done by making a particular Agent for each user or with a call forward script. It is quite possible though. - web based admin - Asterisk itself should 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 could not be a problem as long as you have the bandwidth to handle all the calls. - Voicemail distro lists are easy. Make a particular extension that dials like VoiceMail (mailbox1&mailbox2&mailbox3) etc. Asterisk supports MOH from mp3 files or other audio files. You should change MOH classes per-channel, per-user, or per everything else. Each MOH class is a folder with file(s) in it. You should make as many classes as you want. - Paging is additionally easy. If you have a particular Asterisk server onsite, hook it is sound card up to the paging system. Otherwise you should page through phones (most phones support intercom/paging). You should page through a particular overhead system using either the sound card, or something more specialized - you should obtain paging controllers with a POTS interface (hook it up to a particular ATA (analog telephony adapter, ethernet on 1 side, FXS (station) POTS port on the other), or you should additionally use a VoIP phone to interface a paging system. Grandstream GXP2000 phones for example have a 3.5mm jack which should be easily wired up to a paging system. - Afterhours is easy. Have a particular audio file which contains the after hours message for a location. Dialing a certain extension records or lets you change the file or turn it on/off (you should script this) - For a backup, use a PRI (T1). Probably run the to your central place. Alternatively, you should obtain PRIs to local servers, including local calls will go out including incoming come in the way, with LD calls going out via VoIP. Remember, VoIP PBX does not necessarily mean VoIP phone service. 9. A T1 carries 28 lines. With that will be something you will have to sit down including talk regarding including go over call volume in a pre-sales meeting. Base your decison on that input. If it were me, I will go with a more integrated approach. Install hybrid systems at each location including network them together with VoIP. If the PRI is down between a warehouse including your main office, you are not totally dead. The warehouse still has local lines including should make including receive calls. Bottom line....despite the Asterisk slant provided above...you may simplify the direction the answers come from to this: 1) Forget regarding a hosted solution. 2) Make sure your purchase a PBX that has robust ARS. 3) Find a dealer that is experienced with networked systems.
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