How To Solve A Consolidation Of Your Business
Phone System Needs With An Asterisk VoIP Solution
By Michael Lemm
Here's the scenario.......
You're 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 (~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, 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, initial setup, 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, 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, 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.
- Afterhours is easy. Have an audio file which contains the after hours message for
a location. Dialing a certain extension records or lets you change this file or turn it
on/off (you can script this)
- For a backup, use a PRI (T1). Probably run this to your central place.
Alternatively, you can get PRIs to local servers, and local calls would go out and
incoming come in this way, with LD calls going out via VoIP. Remember, VoIP PBX
doesn't necessarily mean VoIP phone service.
9. A T1 carries 28 lines. With that would be something you would have to sit down
and talk about and go over call volume in a pre-sales meeting. Base your decision
on that input.
If it were me, I would go with a more integrated approach. Install hybrid systems at
each location and network them together with VoIP. If the PRI is down between a
warehouse and your main office, you're not totally dead. The warehouse still has
local lines and can make and receive calls.
Bottom line....despite the Asterisk slant provided above...you may simplify the
direction the answers come from to this:
1) Forget about a hosted solution.
2) Make sure your purchase a PBX that has robust ARS.
3) Find a dealer that's experienced with networked systems.
Michael is the owner of FreedomFire Communications....including
DS3-Bandwidth.com and Business-VoIP-Solution.com. Michael also
authors Broadband Nation where you're always welcome to drop in and catch up
on the latest BroadBand news, tips, insights, and ramblings for the masses.
Phone System Needs With An Asterisk VoIP Solution
By Michael Lemm
Here's the scenario.......
You're 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 (~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, 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, initial setup, 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, 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, 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.
- Afterhours is easy. Have an audio file which contains the after hours message for
a location. Dialing a certain extension records or lets you change this file or turn it
on/off (you can script this)
- For a backup, use a PRI (T1). Probably run this to your central place.
Alternatively, you can get PRIs to local servers, and local calls would go out and
incoming come in this way, with LD calls going out via VoIP. Remember, VoIP PBX
doesn't necessarily mean VoIP phone service.
9. A T1 carries 28 lines. With that would be something you would have to sit down
and talk about and go over call volume in a pre-sales meeting. Base your decision
on that input.
If it were me, I would go with a more integrated approach. Install hybrid systems at
each location and network them together with VoIP. If the PRI is down between a
warehouse and your main office, you're not totally dead. The warehouse still has
local lines and can make and receive calls.
Bottom line....despite the Asterisk slant provided above...you may simplify the
direction the answers come from to this:
1) Forget about a hosted solution.
2) Make sure your purchase a PBX that has robust ARS.
3) Find a dealer that's experienced with networked systems.
Michael is the owner of FreedomFire Communications....including
DS3-Bandwidth.com and Business-VoIP-Solution.com. Michael also
authors Broadband Nation where you're always welcome to drop in and catch up
on the latest BroadBand news, tips, insights, and ramblings for the masses.
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