[sr-dev] Interconnecting SIP servers without the PSTN

greetings all: I have long believed that VOIP and SIP will not reach their full potential until SIP servers can route calls to other SIP servers without having to go through the ancient telephone system, and pay their tolls. There is nothing of substance preventing any SIP server from calling numbers at any other SIP server. They just need to know which numbers are hosted on which servers. There have been several attempts to resolve this issue: freenum.org, e164,org, Dundi (for asterisk). All appear to be dead at this time. I think that one of the reasons for these failures was that all of these systems relied on the public DNS system to exchange server location info. Putting your SIP server address on a public system and advertising that this is the IP of a SIP server is simply begging for hackers to attempt to breach your SIP server. Its like painting a big target on your back. We at Xantek have been working on an alternate approach, using AGI calls and responses to identify routing info. This approach allows us to limit server identification to registered users of the system, and registered users will have to provide identification (something that hackers probably won't do). We also are incorporating a PIN number into the dial string, so that recipients are aware that the call is coming from a valid user. The PIN can be easily changed if fraudulent activity is suspected. We have a working model for Asterisk set up (see voipconnect.tel for details), but we would like to expand into the Kamailio-verse. What we need is a few Kamailio experts to help with the development of the system on Kamailio. If you have any interest in helping, please reply to this post. TIA, Bill