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&amp;T for example) and this should work fine with SIP.<br>
<br>-Vladimir Rodionov <br> <br><br><div class="gmail_quote">On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <span dir="ltr">&lt;<a href="mailto:pjintheusa@gmail.com">pjintheusa@gmail.com</a>&gt;</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 class="h5"><br>
<br>
<br>
On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov&lt;<a href="mailto:vladrodionov@gmail.com">vladrodionov@gmail.com</a>&gt; wrote:<br>
&gt; Good morning,<br>
&gt; This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes<br>
&gt; through PSTN Gateway (1) to freeSWITCH application server (AS) (2).<br>
&gt; AS does some logic and transfers call (or forward) out of Voip provider<br>
&gt; network to another PSTN number User2.<br>
&gt;<br>
&gt;<br>
&gt; This is call bridge<br>
&gt;<br>
&gt;<br>
&gt;     UA1     (PSTN) -                                 -&gt;  UA2 (PSTN)<br>
&gt;               -                                            -<br>
&gt;                 -  (1)                                   -  (4)<br>
&gt;                   -&gt;         PSTN Gateway-&gt;<br>
&gt;                             -                        -<br>
&gt;                        (2) -                        - (3)<br>
&gt;                           -&gt; FreeSWITCH -&gt;<br>
&gt;<br>
&gt;<br>
&gt; This is what I want to acomplish<br>
&gt;                                     (4)<br>
&gt;     UA1     (PSTN) ------------------------------- -&gt;  UA2 (PSTN)<br>
&gt;                     -<br>
&gt;                        -   (1)<br>
&gt;                         -&gt;  PSTN Gateway-&gt;<br>
&gt;                              -                        -<br>
&gt;                        (2)  -                        - (3)<br>
&gt;                             -&gt; FreeSWITCH -&gt;<br>
&gt;<br>
&gt;<br>
&gt; 1. Can it be implemented in FreeSWITCH?<br>
&gt; 2. Does anybody know Voip providers which support out of network call<br>
&gt; transfer/forwarding to PSTN?<br>
&gt;<br>
&gt; TIA<br>
&gt;<br>
&gt; -Vladimir Rodionov<br>
&gt;<br>
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&gt;<br>
&gt;<br>
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</blockquote></div><br>