asterisk users - Jan 2005 - setup questions- many users, little use

Hello All,

I?m on the technology committee for a fraternity at the University of Illinois.
We?re looking into moving from our current ?party line? (one line shared 
between every two rooms) system to a PBX with voicemail in an effort to lower 
our monthly phone bill and provide better communication services.  We?ve 
pretty much settled on Asterisk as we do not wish to rewire all of our pots
lines
and can?t justify $19,000 for Cisco Call Manager.  We do not have many 
incoming/outgoing calls because most people are using their cell phones, but 
we do have to provide local service for 41 rooms, plus common areas and 
possibly remote users.  

Right now our setup is looking as follows:
12 ch T1 with 60 or 80 DID?s (using Digium T100P)
P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk)
41 Sipura ATA?s (SPA-1001)
3 Cisco 7960?s (in common areas, mostly to look good, but also to provide 
directory info)

Our usage/requirements are as follows:
Callers to main number will be greeted with IVR providing directory general 
announcements
Members assigned one of DID numbers which will follow them their entire time 
living in, calls to DID numbers go strait through
Voicemail for all users, as well as several general mailboxes
Call groups based on committees (philanthropy, exec, alumni relations, etc)
Call forwarding (only to internal extensions)
Overhead paging/intercom
Possible remote extensions for members living out of house (using 7960/40?s)
Management/configuration through our current portal system (web based, most 
likely we?ll write this from scratch as we have pretty specific uses)
Possible wake-up call scheduling
Possible configuration from database already storing member information
Future long distance service to Chicago area through collocation or similar

I would greatly appreciate any input as to specific configurations, things to 
watch out for and consider, and any other useful information.  Will our 
equipment selections work well with Asterisk?  Will there be any compatibility 
issues with the T100P and the T1 from SBC?  And what kind of reliability can we 
expect from this setup? Also if anyone has a setup similar to this please let me
know how it worked out.  We will most likely publish a case study, specific 
configuration guide, and extensive documentation after we finish 
implementation, as other fraternities and sororities on campus have also 
expressed interest in our approach to IT management.  

Thanks in advance for any help,
Bill Lattner

Bill Lattner wrote:
>Right now our setup is looking as follows:
>12 ch T1 with 60 or 80 DID?s (using Digium T100P)
>P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk)
>41 Sipura ATA?s (SPA-1001)
>
41 Sipura ATA's? Wow. that's a lot.. Have you thought about using 
channel banks or the Mediatrix product? Of course, if this is 1 ata per 
room, that's a different story. Just thinking that you could install the 
mediatrix in the building's closets and utilize the existing POTs cabling..
Just a thought.. 41 ATAs might be a pain to manage unless you are setup 
and prepared for that..
-Brett

Just a thought SPA-2100 will give you 2 ports and a lan port per ata. 

Jeff

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Bill
Lattner
Sent: Wednesday, January 26, 2005 6:25 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] setup questions- many users, little use


Hello All,

I?m on the technology committee for a fraternity at the University of Illinois.
We?re looking into moving from our current ?party line? (one line shared 
between every two rooms) system to a PBX with voicemail in an effort to lower 
our monthly phone bill and provide better communication services.  We?ve 
pretty much settled on Asterisk as we do not wish to rewire all of our pots
lines
and can?t justify $19,000 for Cisco Call Manager.  We do not have many 
incoming/outgoing calls because most people are using their cell phones, but 
we do have to provide local service for 41 rooms, plus common areas and 
possibly remote users.  

Right now our setup is looking as follows:
12 ch T1 with 60 or 80 DID?s (using Digium T100P)
P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk)
41 Sipura ATA?s (SPA-1001)
3 Cisco 7960?s (in common areas, mostly to look good, but also to provide 
directory info)

Our usage/requirements are as follows:
Callers to main number will be greeted with IVR providing directory general 
announcements
Members assigned one of DID numbers which will follow them their entire time 
living in, calls to DID numbers go strait through
Voicemail for all users, as well as several general mailboxes
Call groups based on committees (philanthropy, exec, alumni relations, etc)
Call forwarding (only to internal extensions)
Overhead paging/intercom
Possible remote extensions for members living out of house (using 7960/40?s)
Management/configuration through our current portal system (web based, most 
likely we?ll write this from scratch as we have pretty specific uses)
Possible wake-up call scheduling
Possible configuration from database already storing member information
Future long distance service to Chicago area through collocation or similar

I would greatly appreciate any input as to specific configurations, things to 
watch out for and consider, and any other useful information.  Will our 
equipment selections work well with Asterisk?  Will there be any compatibility 
issues with the T100P and the T1 from SBC?  And what kind of reliability can we 
expect from this setup? Also if anyone has a setup similar to this please let me
know how it worked out.  We will most likely publish a case study, specific 
configuration guide, and extensive documentation after we finish 
implementation, as other fraternities and sororities on campus have also 
expressed interest in our approach to IT management.  

Thanks in advance for any help,
Bill Lattner

Thanks for your question!

We are also in the process to move our existing telephone system to an asterisk
based solution. Our
setup would look similar, except that the asterisk box would be connected to an
ADSL line for VOIP
_and_ three PSTN lines (which we already use). Otherwise, setup is similar with
about 40 telephones.

When we first thought about the move we considered an only-IP-phone solution.
However, after some
rethinking we moved away from that concept. The most likely end scenario will
now be that we leave
the old telephone wiring in place and connect that to a channel bank or voip
gateway into the
asterisk machine. For the fxo connectivity we would do the same  - connect the
PSTN lines to some
unit (a bank of three Sipura-3000 seems best at the moment).


I suspect that our initial IP-phone-only concept would be similar to the
scenario you presented, as
you would like to move over to use ATAs to connect the analog phones. I suspect
that you would like
to use ATAs because you would like to move away from the 'party line'
arrangement and instead let
all phones have their own line each.

I would strongly advise against this solution for the following reasons:

* You have to put in an entire new cat-5 network wich means putting in cabling,
   switches / hubs, 19 inch cabinets(?) etc.

* You will have a lot of setup efforts at the beginning (each ATA needs its own
   identity for registration purposes [passwords / user names etc]) and ongoing
   administration efforts as your network grows.

* You have security issues, as the SIP protocol is inherently unsafe. As
examples see
         http://www.cert.org/advisories/CA-2003-06.html
         http://www.ee.oulu.fi/research/ouspg/protos
             with
            
http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/index.html
                 for SIP

     or do a google search with the keywords
         SIP vulnerability

     The reason I mention that is that using 41 ATAs (which use the SIP
protocol)
     is _not_ a good idea if the SIP protocol isn't safe. The same problem
_might_
     occur if you use iax adapters, but I don't know anything about security
on that
     area, so pls don't stone me for mentioning this :)


* You could end up with misuse by the users, for they simply can plug in their
computer
   into the voip network and use it for internet browsing or other activities.
Again - lots
   of admin to do on the firewall side. And sysmins are already stressed people
:)

* You have to power the ATAs over the mains which means you have 41 power
supplies
   constantly in the mains. This adds to the fire risk. Additionally there is
the
   potential issue of ATAs breaking if and when they are powered up and for some
reason
   disconnected while downloading new firmware during power-up (not extremely
likely,
   but can happen).

* You have high initial costs of purchase for the ATAs - 41 times ??.?? dollars



Instead I would advise you to go for reuse wherever possible: Use the old
telephone wiring for the
existing telephones and connect it into a channel bank. The Carrier Access Adit
600 has excellent
recommendations if it comes to interoperability with Asterisk, see

http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank

and look at the channel bank checklist
http://www.voip-info.org/wiki-Asterisk+hardware+channel+bank+check

as well before buying a channel bank.

Re-using the old wiring would be an excellent use of the existing resources and
save you a *lot* of
admin and security headaches. On top of it you can connect any standard analog
phone to the system.

To provide the voip functionality you can connect the asterisk box into a dsl
line (ADSL
recommended), and I advise you do that via a firewall. For the firewall, IPcop
comes to mind, it is
free and can be easily installed on any standard PC. See

     http://www.ipcop.org

for further reference. IPcop is a setup-once-and-forget-it system. It just runs
and runs. It has a
web based admin interface and ssh access as well.

If you have an (A)DSL line, use an ADSL router. There are numerous products
around, such as the
D-link DSL504T. A router gives you additional security because of its NAT
facilities.

Overall, design the entire system as modular as possible. For the analog
connectivity, use the
channel bank.  For any outside line use separate devices that convert to
ethernet based protocols.
Any ethernet connections are done through switches/hubs. Having the analog side
separated from the
server side seems to be a much cleaner way of implementation than (for example)
putting cards into
the server and connecting analog lines directly into that.

Using the configuration described above would give you a reliable system with
low initial costs and
low maintenance later.

I hope this helps!

Peter

----

glossary of more obscure terminology - I realize we tend to bash ppl with
acronyms, because this
makes us look professional:

PSTN:    Public switched telephone network. the world's collection of
interconnected voice-oriented
public telephone networks, both commercial and government-owned. It's also
referred to as the Plain
Old Telephone Service (POTS).
   source:
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci214316,00.html

fxo/fxs: What is FXO and FXS? (12 Apr. 2004) expertAnswerCenter.com Ask The
Expert: What is FXO and
FXS? FXO stands for Foreign eXchange Office: simply put, this interface connects
to the analog PSTN
line coming from the central office. FXS stands for Foreign eXchange Station:
simply put, this
interface connects to devices such as analog phones and fax machines.
   source: http://whatis.techtarget.com/wsearchResults/0,,sid9,00.html?query=fxo


ATA:     Analog telephone adapter










> Hello All,
> 
> I?m on the technology committee for a fraternity at the University of
Illinois.
> We?re looking into moving from our current ?party line? (one line shared 
> between every two rooms) system to a PBX with voicemail in an effort to
lower
> our monthly phone bill and provide better communication services.  We?ve 
> pretty much settled on Asterisk as we do not wish to rewire all of our pots
lines
> and can?t justify $19,000 for Cisco Call Manager.  We do not have many 
> incoming/outgoing calls because most people are using their cell phones,
but
> we do have to provide local service for 41 rooms, plus common areas and 
> possibly remote users.  
> 
> Right now our setup is looking as follows:
> 12 ch T1 with 60 or 80 DID?s (using Digium T100P)
> P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk)
> 41 Sipura ATA?s (SPA-1001)
> 3 Cisco 7960?s (in common areas, mostly to look good, but also to provide 
> directory info)
> 
> Our usage/requirements are as follows:
> Callers to main number will be greeted with IVR providing directory general
> announcements
> Members assigned one of DID numbers which will follow them their entire
time
> living in, calls to DID numbers go strait through
> Voicemail for all users, as well as several general mailboxes
> Call groups based on committees (philanthropy, exec, alumni relations, etc)
> Call forwarding (only to internal extensions)
> Overhead paging/intercom
> Possible remote extensions for members living out of house (using
7960/40?s)
> Management/configuration through our current portal system (web based, most
> likely we?ll write this from scratch as we have pretty specific uses)
> Possible wake-up call scheduling
> Possible configuration from database already storing member information
> Future long distance service to Chicago area through collocation or similar
> 
> I would greatly appreciate any input as to specific configurations, things
to
> watch out for and consider, and any other useful information.  Will our 
> equipment selections work well with Asterisk?  Will there be any
compatibility
> issues with the T100P and the T1 from SBC?  And what kind of reliability
can we
> expect from this setup? Also if anyone has a setup similar to this please
let me
> know how it worked out.  We will most likely publish a case study, specific
> configuration guide, and extensive documentation after we finish 
> implementation, as other fraternities and sororities on campus have also 
> expressed interest in our approach to IT management.  
> 
> Thanks in advance for any help,
> Bill Lattner

-- 
dyslexics of the world - untie !