[OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
bogdan at voice-system.ro
Thu Oct 22 21:17:47 CEST 2009
Iñaki Baz Castillo wrote:
> El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
>> 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.
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
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