Deaf folks, Actually this is my first post here, so sorry for any inconvenience. Im planning for a solution a bit larger in scale than ususal. I'm goin to use * as a PSTN gateway with E1 links and use two other 3rd party Gateways for FXO lines. I should be able to switch from every incoming channel to any outgoing one and also to some SIP softphones. I planned to use SER as a sip server but really dont know were I should enforce my call routing mechanisms. Is SER applicable of doing that or should i write any application on the SER to do so ro is there any need for a softswitch at all? Or as a more basical question is there any need for SER, Asterisk cant do it itself?
Any help and hints would be highly appreciated, M. Shokuie Nia. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
