On Apr 23, 2014, at 5:58 AM, Antonio Querubin
<tony(a)lavanauts.org> wrote:
(Please trim inclusions from previous messages)
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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(a)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(a)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(a)lavanauts.org
xmpp: antonioquerubin(a)gmail.com
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