[Asterisk-Users] Minimum budget question ...

[Michael Kane]

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" <andy at beagles-den.demon.co.uk>
To: <asterisk-users at lists.digium.com>
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" <andy at beagles-den.demon.co.uk>
> >To: <asterisk-users at lists.digium.com>
> >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 tim.mcqueen at qualisys.biz 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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