[asterisk-users] Learn some terminalogy before mounting this task.

[James R. Stevens]

All,

I have done research on VoIP for some time now. I'm a Cisco certified
Network Engineer however Telecom is not my strongest suit. I've been a
part of this mailing list for sometime but my delusions of grandeur in
migrating our 25 year old phone system to a new platform have been on
the back burner, until now. I have found my company is moving to a new
location(building) and this provides perfect opportunity.

 

Long story short, I'm very Linux savvy having no problems compiling,
building, making etc.. However getting connected to the PSTN is puzzling
me. My vocabulary is lacking and I need to call our provider this week
and get some circuits moved. So, my confusion....

 

(Current Setup)

We have a T1 coming into the building(FYI-Our Voice and Data are on
separate T's) terminating at the Smart Jack. Then a cable from the T
card(SmartCard) to the channel bank. From the Channel bank lines are
punched down to the block. Then those ANALOG lines are fed into our Big
hunking PBX mounted on the wall and two (Looks to be Rj11) lines come
from it into our VM server.

 

Q&A..

 

We are going to leave the telecom hardware behind.. I want to replace it
all with an Asterisk or Tribox solution.

 

I can tell you our current phone system can handle 7 phone calls at a
time:

   Does this mean the T only has 7 channels provisioned out of the 24
possible?

 

  Does a channel (In terms of the T1) = a port?

 

  How many phone calls can one TDM400 support concurrently? (four ??)

 

  Would I be better off getting a Zapata T1 card and forgetting the
Channel bank all together(Use the digital signal)?

 

  Or Have a channel bank installed and use an OpenSwitch12 solution
working with the analog signal?

 

 If we go with a Zapata T1 card for the Asterisk server would we be able
to provision an analog phone line, for say a FAX machine from it?

 

(Further information)

Ultimately this is the FIRST of 3 offices that will be migrated. This
first office has 30 physical phones and currently no DID (Hoping to
change that as well) Once the second and third offices migrate(about the
same number of stations) we would connect the 3 Asterisk systems over
the WAN giving us free long distance between offices.  The toughest
challenge I foresee is getting overhead intercom/paging to work(We must
be able to hail our warehouse staff over head), also Music on Hold... If
I use Cisco 7940's (SIP) (Or the like) I'd like to integrate our phone
extensions for a 3 offices into the directory

Does this sound do-able?

 

 


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