[sr-dev] Interconnecting SIP servers without the PSTN
bill at telopar.net
Fri Jun 18 18:02:03 CEST 2021
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
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
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