[Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN

[Phillip Jones]

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 at 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
>
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