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.
