[Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
Vladimir Rodionov
vladrodionov at gmail.com
Sat Aug 8 11:39:13 PDT 2009
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 at 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 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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> >
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