maybe look at using something like mod_easyroute. You setup a database of mappings between DID and endpoints. It is designed for the situation you describe.
Hello all,
Is there a way I can dial at number at a registered extension.
Basically what I'd like to do is use Freeswitch as a SBC for a number
of asterisk machines and I'd like to pass an incoming sip invite to a
specific number at each one. Right now I'm specifying the IP in the
bridge command like this sofia/default/NNNNNNNNNNN@xx.xx.xx.xx but if
I dial the registered user directly it dials the contact address at
the registered user and does not pass the inbound number. Basically
what I'm trying to do is this:
pstn -> ITSP -> freeswitch -> asterisk (1-n)
Thanks,
Spencer
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