<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; CHARSET=UTF-8">
<META NAME="GENERATOR" CONTENT="GtkHTML/3.28.1">
</HEAD>
<BODY>
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>
<BR>
<A HREF="http://www.patton.com/products/pe_products.asp?category=406">http://www.patton.com/products/pe_products.asp?category=406</A><BR>
<BR>
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<BR>
<BR>
<BR>
On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:<BR>
<BLOCKQUOTE TYPE=CITE>
Hi,
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
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.
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
PSTN integration:
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
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.
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
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?
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
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?
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
Real Deployments:
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
Supposing I'm to do a real deployment for a client. What are the options that I have for hardware?
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
0) Get IP phones that talk SIP? Is this the most expensive option?
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
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?
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
Thanks
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<BR>
<BR>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
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.
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<PRE>
_______________________________________________
FreeSWITCH-users mailing list
<A HREF="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</A>
<A HREF="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A>
UNSUBSCRIBE:<A HREF="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</A>
<A HREF="http://www.freeswitch.org">http://www.freeswitch.org</A>
</PRE>
</BLOCKQUOTE>
<BR>
</BODY>
</HTML>