[Freeswitch-users] PSTN Integration and real deployments
Campbell Steven
casteven at gmail.com
Thu Apr 1 23:24:15 PDT 2010
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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