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
>
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