[OpenSER-Users-ES] Re: Asterisk y LoopBack en SIP (dolor de cabeza)
Iñaki Baz Castillo
ibc at aliax.net
Mon Aug 27 18:12:34 CEST 2007
El Lunes, 27 de Agosto de 2007, Iñaki Baz Castillo escribió:
> En fin, me gustaría preguntaros por vuestras experiencias exprimiendo el
> stack SIP de Asterisk. ¿Por qué demonios me permite loopback directo pero
> no lo permite si es un alias? ¿acaso se fija en el "To:" por alguna razón?
Leo en:
http://www.voip-info.org/wiki/view/Asterisk+at+large
"I don't think that Asterisk is quite ready to support all live
deployment scenarios that include a 3rd party SIP proxy.
One problem I ran into was Asterisk does not handle looped back calls.
For example a call comes in over PSTN to Asterisk, Asterisk forwards to
your SIP registrar proxy, Registrar does a lookup on the SIP address and
finds that the user is register'd to an analogue phone.
If the SIP registrar redirected using a 3xx response the * will play
along happily, but if the proxy wishes to stay in the loop (maybe you
have a billing application running on it) it would add a Record-Route
header to the SIP request , to say it wishes to receive all subsequent
messages for this call, and then proxy back to the *. The * will ignore
this INVITE totally.
If the user had been registered to a proper SIP end point then the loop
back wouldn't have happened and this works a treat."
Pero realmente NO es mi caso puesto que mi OpenSer no le responde a Asterisk
con un redirect, simplemente OpenSer hace un append_branch y modifica el URI
actual para que sea una extensión del Asterisk.
--
Iñaki Baz Castillo
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