[Freeswitch-users] PSTN Integration and real deployments

[Nandy Dagondon]

IP Channel Bank is new to me. i thought, all the while, that channel banks
uses expensive  E1/T1 interface.  now, it widens my options. tks for this
info.


On Fri, Apr 2, 2010 at 2:24 PM, Campbell Steven <casteven at gmail.com> wrote:

>  Hi Guru,
>
> If you have a bunch of analog extensions from say an old PBX installation
> that you want to reuse (I *think* this is what you are getting at?) then you
> can use a channelbank like these:
>
> http://www.patton.com/products/pe_products.asp?category=406
>
> Which essentially are an up to 32 port ATA, like Nandy says, an ATA's
> purpose is to allow you to use an analogue handset on a VoIP system. This
> would allow you to have each individual analogue extension registering to
> it's own SIP account.
>
> Campbell
>
>
>
> On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:
>
> Hi,
>
>  After long nights and lots of coffee =) , I think I've largely understood
> FreeSwitch. I've been playing with it and have managed most fancy things it
> can do. But I've done this on my LAN using SIP softphones. Here's my problem
> now, I know nothing about PSTN integration and real deployments. Here are my
> questions, mostly based on what I read on wikipedia.
>
>
>
>  PSTN integration:
>
>
>
>  I have an ADSL internet connection, with a split-box? installed by my ISP
> which splits the incoming line to two, one for the phone provided and one
> for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
> also be able to outbound calls to PSTN/VoIP phones via an SIP client
> registered with my FS server through an external gateway or the PSTN line.
>
>
>
>  0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
>
>
>
>  1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal PSTN
> numbers? If yes, where will the number reside, as in will PSTN numbers
> calling me be charged extra?
>
>
>
>  Real Deployments:
>
>
>
>  Supposing I'm to do a real deployment for a client. What are the options
> that I have for hardware?
>
>
>
>  0) Get IP phones that talk SIP? Is this the most expensive option?
>
>
>
>  1) Suppose the client has a traditional plain intercom installment(think
> hotels etc). with phones connecting via RJ11 connectors. Is it possible to
> have something like an ATA with lots of ports working as a hub/switch, So
> that all phones can be plugged into ATA and managed via FS?
>
>
>
>  Thanks
>
>
>
>  PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more with
> it.
>
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