Iñaki Baz Castillo wrote:
El Miércoles, 12 de Agosto de 2009, Alex Balashov escribió:
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.
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.h...