That looks promising, thanks. I was hoping to have FS do it automatically. I.e. inbound dids are assigned to specific asterisk boxes, ITSP dials inbound did of freeswitch, freeswitch dials inbound did @ whereever the registration came from instead of having the IP address hard coded into a route. On Mar 14, 2010, at 2:12 PM, Rupa Schomaker wrote:
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.
On Sun, Mar 14, 2010 at 3:13 PM, Spencer Thomason <spencer@5ninesolutions.com
wrote: 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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