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<vladrodio...@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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
