[OpenSER-Users] Avoid Loops between Openser and Asterisk

[Iñaki Baz Castillo]

2008/4/12, Adrian A <adrianvoip at gmail.com>:
> 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 at aliax.net>