[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:

"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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