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<[email protected]> 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 <[email protected]> > 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 <[email protected]> >> 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<[email protected]> >>> 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 >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > [email protected] >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> [email protected] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
