[OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE

[Brett Nemeroff]

Personally,I think broken UACs should behave like broken UACs. If you start
making exceptions, then they don't get fixed.
Things that are "rigged" to make them appear to work turn into problems that
are hard to detect. or become ignored until they become larger problems.

Like Bogdan said, just because you can, doesn't mean you should. :)

-Brett


On Thu, Oct 22, 2009 at 2:17 PM, Bogdan-Andrei Iancu <bogdan at voice-system.ro
> wrote:

> Iñaki Baz Castillo wrote:
> > El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
> >
> >> Hi,
> >>
> >> After reading lots of docs and mailing lists, it looks like there is now
> >> solution for asterisk available and looks like that might be a long way
> >> till then.
> >>
> >
> > If you mean the link below take into account that it's chan_sip3 which is
> not
> > implemented in asterisk at all since nobody wants to support Olee to do
> that.
> > Asterisk is not interested in SIP.
> >
> >
> >
> >> Maybe its possible to implement that feature in TM?
> >>
> >
> > I really expect a proxy shouldn't behave as a UAC.
> >
> Inaki, Uwe,
>
> Such a feature is possible to do in TM (technically speaking) , but my
> doubt is if this is the correct thing to do - because as you said, more
> or less is not the job of a proxy to sort out such situation (even if it
> can ;) ).
>
> So the question actually is: do we want to be rigorous about what we
> should or should not do, or we want to add some extra options to to help
> with interoperability of some broken/stupid entities??
>
> Blue pill ? Red pill ?? :D
>
> Regards,
> Bogdan
>
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