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.<br>
<br><br><div class="gmail_quote">On Fri, Apr 2, 2010 at 2:24 PM, Campbell Steven <span dir="ltr"><<a href="mailto:casteven@gmail.com">casteven@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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Hi Guru,<br>
<br>
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:<br>
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<a href="http://www.patton.com/products/pe_products.asp?category=406" target="_blank">http://www.patton.com/products/pe_products.asp?category=406</a><br>
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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.<br>
<br>
Campbell<div><div></div><div class="h5"><br>
<br>
<br>
On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:<br>
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Hi,
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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.
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PSTN integration:
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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.
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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?
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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?
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Real Deployments:
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Supposing I'm to do a real deployment for a client. What are the options that I have for hardware?
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0) Get IP phones that talk SIP? Is this the most expensive option?
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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?
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Thanks
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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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