On Apr 23, 2014, at 5:58 AM, Antonio Querubin tony@lavanauts.org wrote:
(Please trim inclusions from previous messages) _______________________________________________
On Wed, 23 Apr 2014, Danny Messano wrote:
necessary, obviously, for the call to be delivered. But, I don't need a SRV record to tell the world I am hosting SIP at ke4rap.ampr.org when that's the de facto destination for a call for ke4rap.
And therein lies the problem. The rest of the SIP world doesn't make that kind of assumption any more so than assuming reaching (123)456-7890 is done by making a SIP call to a de facto destination of 1234567890@1234567890.ampr.org. Seriously?
Unless you want to spend unnecessary time coding this kind of assumption into various SIP software, yeah you do need SRV.
That's ENUM, not DNS SRV, and again, not necessary. I know of very few software SIP clients that don't allow SIP URI dialing. No need for numerics whatsoever. I can dial alphanumeric user@host from any number of clients.
I also addressed the numeric hardware endpoints, such as desk phones and ATAs. I have "translated" numeric dialing to SIP URI dialing for years using the Asterisk dialplan.
exten => 234,1,Dial(SIP/ke4rap@ke4rap.ampr.org).
We don't NEED and should absolutely NOT develop some horrible kludge of a numeric schema for number <> callsign translation. If I have numeric-only endpoints I can address those in MY dialplan. Ultimately on the wire calls should be going peer-to-peer with user@host SIP URI's and only require DNS A and CNAME's for routing.
DM
Antonio Querubin e-mail: tony@lavanauts.org xmpp: antonioquerubin@gmail.com _________________________________________ 44Net mailing list 44Net@hamradio.ucsd.edu http://hamradio.ucsd.edu/mailman/listinfo/44net