Re: [Asterisk-Users] Minimum budget question ...

[denon]

What'd this device set ya back? Have a url?

-d

At 11:45 PM 6/30/2003 -0700, you wrote:

AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with 
SIP up and running with *

Michael Kane wrote:

The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8 FXO.  I've delpoyed hundreds of these IAD's signaling with MGCP.
Not sure if it supports SIP yet.  Hope this helps...
Mike

Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
508-295-2826
----- Original Message ----- From: "Andy Powell" 
<[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 30, 2003 2:51 PM
Subject: Re: [Asterisk-Users] Minimum budget question ...




Hi Tan,

Thanks for the reply. I'll end up asking a load more questions now...

What sort of prices are we talking about for the 24 port
VoIP gateway?
I assume that each port is individually addresable by *?

As I recall the 24 port gateways tend to be terminated at the FXS side
as some 'wierd' connector (wierd in that it's not rj45/11) do you just
wire this to a patch panel?
What codec is in use to get all 24 ports 'running' at the same time..G729?
Does this cause problems since iirc * needs to run in console mode for
the G729 codec to work properly
Thanks for the info... interesting site too :D

Andy



*********** REPLY SEPARATOR  ***********

On 30/06/2003 at 19:21 Tan Aks wrote:



Hi,

We provide asterisk-based solutions to customers based in the uk. One of
our
customers (9 users) is trialling our low-end solution which comprises of
a


box with 2 x X100P (analogue line) cards installed, and a voip carrier

for


outgoing calls. This customer intends to have 13 extensions in his "live"
scenario. The way to use multiple analogue phones is:
      1) get a T100P card and use a T1 channel bank sourced from the US
      2) use a couple of TDM400P cards to give 8 extensions, and use IP
phones for the other extensions
      3) use a voip gateway to provide up to 24 x analogue extensions
per


IP address. VoIP gateways are commonly available and convert analogue

lines


into a SIP/H323 VoIP stream.

You can get an E1 terminated with an RJ45. If you have a coax

termination


then you can use a balun to get rj45 connectivity.

Hope that helps.
Tan (telappliant.com)


----- Original Message ----- From: "Andy Powell" 
<[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 30, 2003 5:26 PM
Subject: RE: [Asterisk-Users] Minimum budget question ...

Tim,

a good comprehensive answer to the question...certainly gave me a few
things
to think about. I do have a few questions though, since I'm in Europe.
Has anyone in Europe set up something equivalent to what Tim suggested?

What sort of prices did it work out at?

How did you solve the channel bank 'issue' in Europe?

I keep reading that E1 lines are coax terminated, is this correct or do

you


usually get a choice from your teleco?

Were there any other issues to contend with?

I'd certainly be interested in the experiences of anyone in Europe...

Thanks

Andy



On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote:



If this is for commercial use, especially if you are going to be selling
this solution, I would suggest that you don't even offer the choice of
analog lines except in the smallest of offices.  Unless you like to
spend a lot of unbillable time supporting them :)

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