hi guru,<br><br>re pstn integration:<br>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. <br>
<br>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.<br><br>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.<br>
<br>re real deployments:<br><br>0) you can mix the client phones - IP hardphones or via FXS gateways (4/8/24 ports) to connect analog phones<br>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<br>
<br>PSTN <==> FS <==> PABX<br><br>i hope it clears up a bit from your cloud confusion :-)<br><br>-nandy<br><br><div class="gmail_quote">On Fri, Apr 2, 2010 at 4:52 AM, guru singh <span dir="ltr"><<a href="mailto:grsingh750@gmail.com">grsingh750@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;">Hi,<div>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.</div>
<div><br></div><div>PSTN integration:</div><div><br></div><div>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.</div>
<div><br></div><div>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?</div>
<div><br></div><div>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?</div>
<div><br></div><div>Real Deployments:</div><div><br></div><div>Supposing I'm to do a real deployment for a client. What are the options that I have for hardware?</div><div><br></div><div>0) Get IP phones that talk SIP? Is this the most expensive option?</div>
<div><br></div><div>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?</div>
<div><br></div><div>Thanks</div><div><br></div><div>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.</div>
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