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

[Iñaki Baz Castillo]

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.

> 
> http://www.codename-pineapple.org/doc/html/sip3_dialog_match.html
> 
> BR
> 
> Uwe
> 
> Iñaki Baz Castillo schrieb:
> > El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
> >>> I don 't understand... first you said that the GW send a call to
> >>> OpenSIPS and OpenSIPS to both AST1 and AST2 (Asterisk). And the problem
> >>> is in the GW because ignores the second 200.
> >>> Am I wrong?
> >>
> >> The setup is:
> >> asterisk(gw) <opensips> ast1+ast2
> >>
> >> No, you are right. So I need to fix that problem on the asterisk(gw) not
> >> on AST1 and AST2.
> >
> > ok ok.
> >
> > I remember that Olle (chan_sip) commented that Asterisk was tested for
> > this scenario (receiving two 200 for INVITE) in a SIPit, but I don't
> > remember the results... :)
> >
> > Of course, you should enable "pedantic=yes" so in this way Asterisk
> > """"is supposed" to match To/From tags also. However I would trust it too
> > much...
> 


-- 
Iñaki Baz Castillo <ibc at aliax.net>