27 Aug
2007
27 Aug
'07
6:12 p.m.
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