Actually, this is what I need <br><br><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect</a><br><br>Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN number?<br>
<br>-Vladimir Rodionov<br><br><br><div class="gmail_quote">On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <span dir="ltr"><<a href="mailto:vladrodionov@gmail.com">vladrodionov@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty sure it is doable because voxeo offers this<br>option for their Voice XML customers but I am not interested in a hosted solution at the time - it is quite expensive. As far as I understood, Voip provider MUST have pstn call transfer feature enabled by telecom provider (AT&T for example) and this should work fine with SIP.<br>
<font color="#888888">
<br>-Vladimir Rodionov <br></font><div><div></div><div class="h5"> <br><br><div class="gmail_quote">On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <span dir="ltr"><<a href="mailto:pjintheusa@gmail.com" target="_blank">pjintheusa@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi there,<br>
<br>
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re<br>
INVITE) and only pass back the media to the network, or pass back<br>
signaling also (SIP REFER)?<br>
<br>
I know several suppliers who support SIP re INVITE but none that<br>
support SIP REFER.<br>
<br>
Check out <a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect" target="_blank">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect</a><br>
and <a href="http://wiki.freeswitch.org/wiki/Bypass_Media" target="_blank">http://wiki.freeswitch.org/wiki/Bypass_Media</a><br>
<div><div></div><div><br>
<br>
<br>
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<<a href="mailto:vladrodionov@gmail.com" target="_blank">vladrodionov@gmail.com</a>> wrote:<br>
> Good morning,<br>
> This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes<br>
> through PSTN Gateway (1) to freeSWITCH application server (AS) (2).<br>
> AS does some logic and transfers call (or forward) out of Voip provider<br>
> network to another PSTN number User2.<br>
><br>
><br>
> This is call bridge<br>
><br>
><br>
> UA1 (PSTN) - -> UA2 (PSTN)<br>
> - -<br>
> - (1) - (4)<br>
> -> PSTN Gateway-><br>
> - -<br>
> (2) - - (3)<br>
> -> FreeSWITCH -><br>
><br>
><br>
> This is what I want to acomplish<br>
> (4)<br>
> UA1 (PSTN) ------------------------------- -> UA2 (PSTN)<br>
> -<br>
> - (1)<br>
> -> PSTN Gateway-><br>
> - -<br>
> (2) - - (3)<br>
> -> FreeSWITCH -><br>
><br>
><br>
> 1. Can it be implemented in FreeSWITCH?<br>
> 2. Does anybody know Voip providers which support out of network call<br>
> transfer/forwarding to PSTN?<br>
><br>
> TIA<br>
><br>
> -Vladimir Rodionov<br>
><br>
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