We need to configure or program our Asterisk (using Free PBX) so when SIP calls comes to Asterisk, the call immediately route to Skype network without checking the extension or DID on the database. We want to avoid the database checking in the Asterisk and send directly to Skype (we already have Digium license).
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hi. can u pls explain more about config? in what form it come to asterisk? as number or as skype name? how it must go to skype? to skypeout or to skype name?