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

[Michael Jerris]

I don't know of any sip carriers who will let you do refer.  you will  
need to find a carrier who supports it.  FreeSWITCH will have no  
problem sending it but I doubt you will find a carrier who will let  
you do it easily.

Mike

On Aug 8, 2009, at 3:12 PM, Vladimir Rodionov wrote:

> 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 at 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 at 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 at 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 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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