Hi,
I saw this link in voip-info wiki site,
http://voip-info.org/wiki-Asterisk+at+large
Quote,
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.
I'd like to know if there is any solution to this problem.
Thanks,
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