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.