[Asterisk-Users] Minimum budget question ...
Ing. Angel Gomez Garcia
angom at telnor.net
Mon Jun 30 23:45:35 MST 2003
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" <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 :)
>>>>
>>>>
>>>>
>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
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>>>
>>>
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>>>
>>>
>>
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