[Freeswitch-users] PSTN Integration and real deployments
guru singh
grsingh750 at gmail.com
Thu Apr 1 13:52:08 PDT 2010
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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