Re: [asterisk-users] Big practical systems

2010-11-08 Thread Cary Fitch


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joel Maslak
Sent: Sunday, November 07, 2010 2:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big practical systems

I believe this looks like a standard channel bank.  Asterisk generates all
audio.  Ring and hook status are sent out of band.  Dial tones are in-band.
Ringback, busy, congestion are in-band audio.  I would think a standard T1
card would be fine.

That said, I would verify this with the LEC. 
===

Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed
bit?

Cary Fitch 


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Big practical systems

2010-11-08 Thread Joel Maslak
On Mon, Nov 8, 2010 at 7:05 AM, Cary Fitch ca...@usawide.net wrote:

 Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed
 bit?



It won't be ISDN.  It will be some form of RBS.  You probably have several
choices as to which type of RBS (probably several ESF options, you'll
probably pick one of them; you may be able to use SF as well).

You should probably work with your LEC to figure out exactly what they will
hand off to you.  You might make a costly mistake if you don't.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Big practical systems

2010-11-07 Thread Benoit
On 07/11/2010 19:29, Cary Fitch wrote:
 I don't want to start the How many calls can Asterisk handle? discussion
 or How many angels can stand on the point of a pin? discussion either.

 But can anyone contribute some practical knowledge of systems that take in
 channel bank T1s or DS3s from far away, and process the calls?

 I am looking for real world, been there, done that, or check the 'Belchfire
 Systems GigaFiber 65536' system.

 Not to start the discussion, but Is there a board that will take a DS3 (672
 channels) and a system that will handle the calls, or is that a silly
 question?

 Is there an IP box that would take the DS3 and then a system that would
 handle the calls? My guess would be yes because the actual call load would
 be far lower than 672 calls.  Maybe 100-150 or so simultaneous.

 Each line/call would have to have absolute caller ID.  In other words, PSTN
 call handling.

 Cary

Hi,

Did you saw this before:
http://lists.digium.com/pipermail/asterisk-users/2008-April/209146.html
?

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Big practical systems

2010-11-07 Thread Andrew Latham
inline

 I don't want to start the How many calls can Asterisk handle? discussion
 or How many angels can stand on the point of a pin? discussion either.

You just did

 But can anyone contribute some practical knowledge of systems that take in
 channel bank T1s or DS3s from far away, and process the calls?

Most of us are far too busy.  These things are best learned the hardway.

 I am looking for real world, been there, done that, or check the 'Belchfire
 Systems GigaFiber 65536' system.

Its called Asterisk.

 Not to start the discussion, but Is there a board that will take a DS3 (672
 channels) and a system that will handle the calls, or is that a silly
 question?

There are many.  The primary problem it getting a provider to provide
you with a DS3.

 Is there an IP box that would take the DS3 and then a system that would
 handle the calls? My guess would be yes because the actual call load would
 be far lower than 672 calls.  Maybe 100-150 or so simultaneous.

There are a few solutions here and several expensive chunks of
hardware. Do you want to put all your eggs in one basket?

 Each line/call would have to have absolute caller ID.  In other words, PSTN
 call handling.

That is between you and the provider.  The technology exists on the wire.

 Cary

Gringo Malvado...

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Big practical systems

2010-11-07 Thread Joel Maslak
I believe this looks like a standard channel bank.  Asterisk generates all 
audio.  Ring and hook status are sent out of band.  Dial tones are in-band.  
Ringback, busy, congestion are in-band audio.  I would think a standard T1 card 
would be fine.

That said, I would verify this with the LEC. 

On Nov 7, 2010, at 1:22 PM, Cary Fitch ca...@usawide.net wrote:

 Alternate question:
 
 Asterisk/PSTN oriented.
 
 If an Asterisk system were interfaced via a T1 to a local telco loop to a
 customer premises:
 
 (This is not a T1 to the customer premises, but a T1 to the telco who then
 demuxes it to copper to the customer premises.  IE. In Telecom terms an
 EEL.)
 
 Will Asterisk handle that scenario with common drivers and cards?
 
 Who generates the customer audio comfort sounds, ringing, busy, etc?
 
 
 
 Cary
 I know a lot, but not everything.
 
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Big practical systems

2010-11-07 Thread David Backeberg
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch ca...@usawide.net wrote:
 But can anyone contribute some practical knowledge of systems that take in
 channel bank T1s or DS3s from far away, and process the calls?

Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into PRIs.

 Not to start the discussion, but Is there a board that will take a DS3 (672
 channels) and a system that will handle the calls, or is that a silly
 question?

If by board, you mean PCI board for shoving in something with an intel
cpu, not that I've ever heard. Digium sells 4x port PRI boards, and
some competitor sells an 8x port PRI board, but I've never tried any
boards not made by Digium.

The only thing silly is the idea of trusting that many calls to PC hardware.

 Is there an IP box that would take the DS3 and then a system that would
 handle the calls?

Yes, embedded hardware from a vendor you've heard of will do that.
Cisco makes a 3845 which can terminate about 20 PRIs in one appliance.

 My guess would be yes because the actual call load would
 be far lower than 672 calls.  Maybe 100-150 or so simultaneous.

Well, then it's not really a DS3. If it can't do the whole thing
without melting down, it shouldn't advertise itself as DS3. The Adtran
gear works rock solid when pushed to the limit.

If you're just talking 150 calls, you could do that with two 4x port
cards in a single PC. I thought you were talking a lot bigger.

 Each line/call would have to have absolute caller ID.  In other words, PSTN
 call handling.

Ummm, there's no such thing as absolute caller ID. You wanna try that
question again? callerID is not legally binding, is not used by
billing, anybody can spoof it.

The closest you can get is to have a LEC provide ANI. You don't need
PRI to get that. You can get that via a quality voip provider, or
yourself using your own termination gear to convert into voip.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Big practical systems

2010-11-07 Thread Cary Fitch

Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into PRIs.

 Thanks, will look at that.  Ah, a DS3/T1 mux.  I was looking for a DS3
PC Card... it would have 672 channels but the system doesn't need to
handle but 20% of them at one time.

If you're just talking 150 calls, you could do that with two 4x port
cards in a single PC. I thought you were talking a lot bigger.

==I mean DS3 with 672 channel paths. There are 672 subscribers out
there.  I am saying that only a percentage of them are talking at peak
times.  We need to supervise 672 lines and expect 15% to talk at the same
time.

 Each line/call would have to have absolute caller ID.  In other words,
PSTN
 call handling.

Ummm, there's no such thing as absolute caller ID. You wanna try that
question again? callerID is not legally binding, is not used by
billing, anybody can spoof it.

===I mean we have to provide service and know what line is calling, not
just provide anonymous service to a lot of people.  We can't just mux a
bunch of lines in to the Asterisk box with no identification.


Cary


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users