12 Apr
2008
12 Apr
'08
1:59 a.m.
2008/4/12, Adrian A <adrianvoip(a)gmail.com>om>:
If the INVITE comes in from Asterisk, OpenSER replies
with 480 and lets
Asterisk deal with sending call to voicemail (e.g. dialstatus =
unavailable). This eliminates the loop because the INVITE does not come back
to Asterisk.
Ok ok, I understand what you mean. But that is just useful for
redirection to voicemail and so.
For example, imagine you have Asterisk as PSTN gateway. Imagine
Asterisk receives a call from PSTN and routes it to OpenSer, and
OpenSer destination user has a redirection to a mobile number. Then
OpenSer would route the invite back to Asterisk who will detect a loop
(an spiral in fact, but Asterisk is buggy here).
Regards.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>