9 Feb
2009
9 Feb
'09
5:28 p.m.
I have two Asterisk boxes, A1 & A2, each running a separate telephony app. We have an external SIP service with DID's NNNNN200 -> NNNNN299. We want to direct the incoming SIP calls so that the DID's NNNNN200 -> NNNNN219 go to Asterisk server A1 and NNNNN220 -> NNNNN299 to Asterisk server A2. Yes we really just want the calls switched on the DID.
Are you thinking about using FreeSWITCH to direct these calls? Something like this? SIP Provider <--> FS <--+--> A1 +--> A2 I just want to make sure that we understand what you are trying to accomplish and why you might need FS in this scenario... -MC