With the risk of beeing flamed from here to eternity.... Eherm.. On the subject of Asterisk-to-Asterisk interchange one word that immediately came to my mind was SS7, or rather SS7oIP.
By implementing a true IN into Asterisk would indeed bring Asterisk into the big league and solve a lot of questions about interoperability and connections to the "real world" of tier one carriers. I am by no means an expert on this subject but I know from a previous projects in Thailand that keeping track of a 6,500+ Cisco AS5300 VoIP network without IN is hard work even on the billing and service activation part where we did our part... :-) Note, by implementing IN into Asterisk (in theory) it doesn't matter what streaming protocol is used, PSTN, SIP, H323, MGCP, IAX or "IP over Avian Carriers with Quality of Service" (RFC 2549), the voice stream can be routed via a different path while call control is still handled by Asterisk. Ok, I've had my saying.. I'll return to lurking mode.. -- Soren ----- Original Message ----- From: "Olle E. Johansson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, April 20, 2004 10:14 PM Subject: Re: [Asterisk-Dev] Transfer of variables to remote servers (new subject) > John Todd wrote: > > more significant and vexing > > problem of transferring values OUT of a particular server to a remote > > server. > > > > I am uncertain how this should be created, actually. How do you hand > > things off in SIP? How about IAX2? MGCP? Zap?(1) How do you read them > > from the other side? How do you refuse them?(2) Can you get a list of > > the attached values at the other end?(3) > > I would suggest we concentrate on Asterisk to asterisk for this functionality. > Which suggest IAX2. > There need to be a "trust" concept - do we trust data from this server, or not? > > > This would be _TREMENDOUSLY_ powerful if it could be well designed and > > had even basic functionality in SIP and IAX2. Think of this very small > > variable name example: "__caller-is-ceo" > For SIP, I can't recall an existing protocol extension. There's a lot of > work going on in the 3G space so maybe there's functionality for this > to be found there - metadata on a call to be handled only between one > provider's trusted servers. > > > /O > _______________________________________________ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > _______________________________________________ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
