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.

-Vladimir Rodionov

On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones <pjintheusa@gmail.com> wrote:

Are your calls coming in on TDM or SIP trunks? Are your calls answered
by FreeSWITCH before you need to redirect them?

On Sat, Aug 8, 2009 at 11:52 AM, Vladimir

Rodionov<vladrodionov@gmail.com> wrote:
> Actually, this is what I need
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>
> Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
> number?
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov@gmail.com>
> wrote:
>>
>> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
>> sure it is doable because voxeo offers this
>> 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.
>>
>> -Vladimir Rodionov
>>
>>
>> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa@gmail.com>
>> wrote:
>>>
>>> Hi there,
>>>
>>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>>> INVITE) and only pass back the media to the network, or pass back
>>> signaling also (SIP REFER)?
>>>
>>> I know several suppliers who support SIP re INVITE but none that
>>> support SIP REFER.
>>>
>>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>>
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov@gmail.com>
>>> wrote:
>>> > Good morning,
>>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
>>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
>>> > AS does some logic and transfers call (or forward) out of Voip provider
>>> > network to another PSTN number User2.
>>> >
>>> >
>>> > This is call bridge
>>> >
>>> >
>>> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
>>> >               -                                            -
>>> >                 -  (1)                                   -  (4)
>>> >                   ->         PSTN Gateway->
>>> >                             -                        -
>>> >                        (2) -                        - (3)
>>> >                           -> FreeSWITCH ->
>>> >
>>> >
>>> > This is what I want to acomplish
>>> >                                     (4)
>>> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
>>> >                     -
>>> >                        -   (1)
>>> >                         ->  PSTN Gateway->
>>> >                              -                        -
>>> >                        (2)  -                        - (3)
>>> >                             -> FreeSWITCH ->
>>> >
>>> >
>>> > 1. Can it be implemented in FreeSWITCH?
>>> > 2. Does anybody know Voip providers which support out of network call
>>> > transfer/forwarding to PSTN?
>>> >
>>> > TIA
>>> >
>>> > -Vladimir Rodionov
>>> >