<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I don't know of any sip carriers who will let you do refer. you will need to find a carrier who supports it. FreeSWITCH will have no problem sending it but I doubt you will find a carrier who will let you do it easily.<div><br></div><div>Mike</div><div><br><div><div>On Aug 8, 2009, at 3:12 PM, Vladimir Rodionov wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">A call is coming on SIP trunk. From PSTN. I does not need to be answered, actually - I need to do some logic before redirecting call but I can answer call as well It won't break the app logic.<br><br>-Vladimir Rodionov<br> <br><div class="gmail_quote">On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones <span dir="ltr"><<a href="mailto:pjintheusa@gmail.com">pjintheusa@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left-width: 1px; border-left-style: solid; border-left-color: rgb(204, 204, 204); margin-top: 0pt; margin-right: 0pt; margin-bottom: 0pt; margin-left: 0.8ex; padding-left: 1ex; position: static; z-index: auto; "> Are your calls coming in on TDM or SIP trunks? Are your calls answered<br> by FreeSWITCH before you need to redirect them?<br> <br> On Sat, Aug 8, 2009 at 11:52 AM, Vladimir<br> <div><div></div><div class="h5">Rodionov<<a href="mailto:vladrodionov@gmail.com">vladrodionov@gmail.com</a>> wrote:<br> > Actually, this is what I need<br> ><br> > <a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect" target="_blank">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<br> > number?<br> ><br> > -Vladimir Rodionov<br> ><br> ><br> > On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <<a href="mailto:vladrodionov@gmail.com">vladrodionov@gmail.com</a>><br> > wrote:<br> >><br> >> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty<br> >> sure it is doable because voxeo offers this<br> >> option for their Voice XML customers but I am not interested in a hosted<br> >> solution at the time - it is quite expensive. As far as I understood, Voip<br> >> provider MUST have pstn call transfer feature enabled by telecom provider<br> >> (AT&T for example) and this should work fine with SIP.<br> >><br> >> -Vladimir Rodionov<br> >><br> >><br> >> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <<a href="mailto:pjintheusa@gmail.com">pjintheusa@gmail.com</a>><br> >> wrote:<br> >>><br> >>> 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> >>><br> >>><br> >>><br> >>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<<a href="mailto:vladrodionov@gmail.com">vladrodionov@gmail.com</a>><br> >>> 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> >>> ></div></div></blockquote></div></blockquote></div></div></body></html>