<div>Hi List,</div>
<div>I'm having a problem with the following setup.</div>
<div> </div>
<div>Asterisk as B2BUA and OpenSIPS as proxy and Registrar, both running on the same Box with different virtuals IPs.</div>
<div> </div>
<div>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).</div>
<div> </div>
<div>Asterisk IP: 10.121.1.251 asterisk16-1.6.0.1</div>
<div>OpenSIPS IP: 10.121.1.252 opensips-1.4.2</div>
<div>Caller IP: 10.121.1.101 Bria 2.2</div>
<div>Called IP: 10.121.1.104 Bria 2.2</div>
<div> </div>
<div>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.</div>
<div>So, Is it possible to change this behavior without modify the source code?</div>
<div> </div>
<div>The following are the relevant fragments of the opensips.cfg file</div>
<div> ------------------------</div>
<div><span lang="EN"># ----- TM params -----<br>modparam("tm", "fr_timer", 10)<br>modparam("tm", "fr_inv_timer", 20)<br>modparam("tm", "wt_timer", 5)<br>modparam("tm", "delete_timer", 2)<br>
modparam("tm", "ruri_matching", 1)<br>modparam("tm", "via1_matching", 1)<br>modparam("tm", "unix_tx_timeout", 2)<br>modparam("tm", "restart_fr_on_each_reply", 1)<br>
modparam("tm", "pass_provisional_replies", 1)<br></span></div>
<div><span lang="EN">#-----Failure Route----</span></div>
<div><span lang="EN">failure_route[1] {<br></span><span lang="EN"></span></div>
<div><span lang="EN"> if (t_was_cancelled()) {<br> exit;<br> }<br>}</span></div>
<div><span lang="EN"></span> </div>
<div><span lang="EN">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.</span></div>
<div><span lang="EN"></span> </div>
<div><span lang="EN">Regards,</span></div>
<div><span lang="EN"></span> </div>
<div><span lang="EN">Ricardo</span></div>