12 Aug
2009
12 Aug
'09
1:54 p.m.
Iñaki Baz Castillo wrote:
El Miércoles, 12 de Agosto de 2009, Alex Balashov
escribió:
Bah, keeping track of all this stuff is way, way too hard. Let's all
switch to P2P2PSIP[1].
I know you said specifically not to refer you to it[2], but, alas.
[1] http://tools.ietf.org/id/draft-kaplan-sip-four-oh-00.txt
[2]
https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-April/019182.…
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
That is quite elegant, and easy to get around in
the Asterisk dial plan
so that routing can still happen on "extension," where "extension"
is
the dialed number:
[generic-incoming]
exten => _.,1,Set(dialed=${SIP_HEADER(P-Dialed-Number)})
exten => _.,n,Goto(incoming-route,${dialed},1)
But in other endpoints it would be quite hard or impossible.
Sure, that's why I told about some RFC's defining such a specification.