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: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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>
>>> ><br>
>>> > _______________________________________________<br>
>>> > FreeSWITCH-users mailing list<br>
>>> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>>> ><br>
>>> ><br>
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>><br>
><br>
><br>
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