[Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

[Wiley Siler]

Sounds like a PRI T1 will be fine for you to start with.  It offers you
23 voice channels (one channel is used for signaling).

That means you can get a single Digium T1 card for around $600 or you
can get a quad T1 card for around $220 (with echo cancellation).  If
there is no move to expand, then get the regular T1 card and save some
cash.  New equipment is always coming out so by the time you are ready
to expand there may be something new out.
Example:
http://www.voipsupply.com/product_info.php?cPath=99_103&products_id=415

The Voice T1 card is and isn't like a normal T1.  I am sure you are
thinking T1 = Internet.  Well, T1 can equal phone channels to.  In this
case, your PRI is delivering voice channels so the card will be your
server's interface to the telco side.  Once the card is connected to the
PRI T1, Asterisk will take care of routing any calls it receives from
your internal users.  It will receive the calls from the internal users
via the NIC in the server.  The NIC acts as the <user> side of this
whole shebang.  Phones or SIP/IAX devices on the same LAN as the NIC of
the server will be able to connect to the NIC via standard TC/IP address
you assign when you build the server.  Protocols are taken care of
automatically, just config the box accordingly.  Dial plans allow users
connected to your PBX to route without the user knowing squat.  The
system knows that number 18001234567 should go to the PRI and it routes
it such.  In essence, aside from being an application server (voicemail,
IVR, etc) Asterisk is also a router.  That being said, it also means
that an * box can split data and voice if configured properly.  Some T1
providers will offer you a split T1 that has 512K data for instance
along with 15 voice channels (1 channel for signaling) or whatever
permutation of the bandwidth you choose.  For ease of use, I would
recommend you stick to just voice over your T1.

So the path of a call looks like this....

User Handset --> Your Network --> NIC on * PBX --> Dial Plan on PBX
Parses --> Sends Call Out of the PRI via the T1 Card

Regarding remote access, there are several ways.  You can allow VPN into
your network then your users can connect just like they were local.  You
can use IAX protocol devices like the IAXy to connect them with an
adapter and a hard phone.  Generally speaking, if you expose a SIP port
to the internet (security caution BTW) then you can have your users
connect from anywhere.  Just remember that there are security issues.
VPN is the best method for security.  

Your numbers will come from whomever your get your PRI from.  MCI or
anyone like that can offer you something.  DIDs are really really cheap
so don't worry too much about that.  Just tell your salesman how many
you want.

T1 lines are Digital.  That says it all.  Better quality of sound
(usually) and more features with more control.  Keeping your 20 lines is
an option of course.  You would need a channel bank and a T1 card.  The
channel bank would accept the analog POTS lines and allow you to connect
your * server to it via a T1 interface.  So that would be.....

POTS Lines --> Channel Bank --> T1 Card on Asterisk Box
Example: http://www.voipsupply.com/product_info.php?products_id=922

There are PCI cards for POTS but they only support 4 lines per card.  By
the time you get to 20 lines your server will be in IRQ hell.  Better
not to deal with it.  The channel bank is a viable solution but like all
other solutions it comes at a cost for hardware.  Ease of setup and the
fact that you do not have to wait 30 days for a T1 install make it a
nice option for some though.

You can have as many DIDs as you want on a digital system line the PRI.
It works like so.  There are 23 channels.  No channel is tied to any one
particular phone number.  The information on what number is being called
is passed to the PRI which passes it along to the T1 card.  So the
channel number in use is irrelevant.  All we need to know is who is
calling and for what number.  The T1 card passes the data to Asterisk
which uses its dialing rules to decide who gets the call.  Maybe the
number in question is support so you send it to a queue.  Maybe it is
for someone's direct line so you sent it to their desk.  The options are
pretty endless.  The only catch is this....  50 DIDs does not equal 50
calls at once.  Something to remember.  Only 23 of the 50 DIDs could
ever possibly be in use at once.  Equally important, how you set your
hunt groups upstream will matter when it comes to line usage.  If you
get a lot of calls on an 800 number, setting all 23 lines as huntable
would leave you with no outward dialing if you got really busy.  That
being the case, you would set a hunt of 20 for the 800 number for
instance and leave 3 out.  That way 3 lines would be available for your
DID pool or for someone to make an outgoing call.   

BTW - I am in no way a telco expert so if I made a mistake, someone on
the list is sure to jump on it and correct me.  They always do... 8)

Cheers,
Wiley

PS.  Since this is getting quite long winded, please feel free to
contact me off list for any clarification or recommendations.  I left
this one on list some anyone could correct any errors I may have made.