[Freeswitch-users] PSTN Integration and real deployments

[Nandy Dagondon]

hi guru,

re pstn integration:
0) get an ATA. connect the FXS port to an analog phone and the FXO port to
your phone line. set the FXO port to dial (send INVITE) to Freeswitch after
1 or 2 rings. you can create an Auto-Attendant on FS to handle this call.
you can control the route of outgoing calls (PSTN or VoIP) via dialplan.

i hv tried Grandstream HT-503 but the FXO port has problems. i'd like to get
Audiocodes. grab their User Manual to give you an idea.

1) you hv to register w/ a VoIP provider if u route your calls via Internet.
the incoming number is usually offered as an option. re charges. it depends
on the plan you get.

re real deployments:

0) you can mix the client phones - IP hardphones or via FXS gateways (4/8/24
ports) to connect analog phones
1) yes. that's multi-port FXS gateways. there are multiport FXO gateways
where you can place FS to handle PSTN calls for the legacy PABX

PSTN <==> FS <==> PABX

i hope it clears up a bit from your cloud confusion :-)

-nandy

On Fri, Apr 2, 2010 at 4:52 AM, guru singh <grsingh750 at gmail.com> 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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