So we’re transitioning from a FXO + SIP backup to a pure SIP solution.
Currently we have a 800-number provider that maps to our PTSN analog phone number. That number rolls over to the SIP trunk if more than 1 call comes in. Our 800-number provider is separate from the local PTSN and the SIP provider.
What we want is for the 800-number to attempt the first SIP provider, but if that provider is down, roll over to a backup SIP provider who is not associated with the first one.
With the analog line, this was easy because if it got a busy or not in service signal, the 800 number hunted to the SIP.
Is it the same process with a pure sip solution? Any advice? I’d like to keep the SIP providers and the 800 number provider separate so that if the primary SIP provider goes down entirely, it doesn’t affect the 800 number.
In the PSTN world Toll-free inbound calls MUST be redirected to an existing regular PSTN number (which might be ultimately a SIP connection) as you are used to and they will only forward the number of “paths” that you pay for, not withstanding all the limitations off forwarding and hunt groups that that “landing” number has in the PSTN world. With voip you have more options, many VSP’s will terminate such inbound calls directly to your server, some provide for “fail-over” routing when you broke it.
There is nothing that Asterisk/FreePBX can do to facilitate that as the failed call will never reach it.
Initially, just talk to your provider and if they don’t support SIP termination directly, consider porting the number(s) and your underlying “RespOrg” to a vsp that does what you ask for, it will generally be cheaper, depending on call volume (deduced by me) expect sub 10 cent per minute calls.