[OpenSIPS-Users] CANCEL and 408 Request Timeout when fr_inv_timer expires

[Ricardo Lopez Camino]

Hi List,
I'm having a problem with the following setup.

Asterisk as B2BUA and OpenSIPS as proxy and Registrar, both running on the
same Box with different virtuals IPs.

Clients are registered in opensips, but the INVITEs are routed to asterisk
(caller leg) and then asterisk generates an INVITE back to opensips (called
leg).

Asterisk IP: 10.121.1.251      asterisk16-1.6.0.1
OpenSIPS IP: 10.121.1.252   opensips-1.4.2
Caller IP: 10.121.1.101          Bria 2.2
Called IP: 10.121.1.104         Bria 2.2

When the fr_iv_timer expires OpenSIPS sends a 408 to asterisk and that event
should trigger the voicemail greeting (200 OK) in the Caller Leg. The
problem is that OpenSIPS sends a CANCEL after the 408 and asterisk can not
send the 200 OK. This is some kind of race condition because sometimes the
200 OK (in the caller leg) is sent but the call is canceled anyway and the
voicemail greeting is not reproduced.
So, Is it possible to change this behavior without modify the source code?

The following are the relevant fragments of the opensips.cfg file
 ------------------------
# ----- TM params -----
modparam("tm", "fr_timer", 10)
modparam("tm", "fr_inv_timer", 20)
modparam("tm", "wt_timer", 5)
modparam("tm", "delete_timer", 2)
modparam("tm", "ruri_matching", 1)
modparam("tm", "via1_matching", 1)
modparam("tm", "unix_tx_timeout", 2)
modparam("tm", "restart_fr_on_each_reply", 1)
modparam("tm", "pass_provisional_replies", 1)
#-----Failure Route----
failure_route[1] {
        if (t_was_cancelled()) {
                exit;
        }
}

I dont want to use Voicemail Diversion (with a new redirected INVITE
generated by opensips), I would rather use the asterisk VoiceMail app as
second priority after the Dial app.

Regards,

Ricardo
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