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
|Time | 10.121.1.101 | 10.121.1.252 | 10.121.1.251 |
|0.000 | INVITE SDP ( telephone-event) |
|SIP From: sip:[email protected] To:sip:[email protected]
| |(1083) ------------------> (5060) | |
|0.068 | | INVITE SDP ( telephone-event)
|SIP Request
| | |(9728) ------------------> (5060) |
|0.441 | | 100 Trying| |SIP
Status
| | |(9728) <------------------ (5060) |
|1.278 | | 180 Ringing |SIP
Status
| | |(9728) <------------------ (5060) |
|1.281 | 180 Ringing | |SIP
Status
| |(1083) <------------------ (5060) | |
|20.862 | 408 Request Timeout | |SIP
Status
| |(1083) <------------------ (5060) | |
|20.863 | | CANCEL | |SIP
Request
| | |(9728) ------------------> (5060) |
|20.863 | | 487 Request Terminated |SIP
Status
| | |(9728) <------------------ (5060) |
|20.863 | | 200 OK | |SIP
Status
| | |(9728) <------------------ (5060) |
|20.869 | | ACK | |SIP
Request
| | |(9728) ------------------> (5060) |
|20.967 | ACK | | |SIP
Request
| |(1083) ------------------> (5060) | |
|Time | 10.121.1.251 | 10.121.1.252 | 10.121.1.104 |
|0.936 | INVITE SDP ( h263-1998) | |SIP
From: sip:[email protected] To:sip:[email protected]
| |(15557) ------------------> (5060) | |
|1.054 | | INVITE SDP ( h263-1998) |SIP
Request
| | |(5060) ------------------> (2829) |
|1.162 | | 100 Trying| |SIP
Status
| | |(5060) <------------------ (2829) |
|1.271 | | 180 Ringing |SIP
Status
| | |(5060) <------------------ (2829) |
|1.275 | 180 Ringing | |SIP
Status
| |(15557) <------------------ (5060) | |
|20.746 | 408 Request Timeout | |SIP
Status
| |(15557) <------------------ (5060) | |
|20.747 | ACK | | |SIP
Request
| |(15557) ------------------> (5060) | |
|20.784 | BYE | | |SIP
Request
| |(15557) ------------------> (5060) | |
|20.855 | | CANCEL | |SIP
Request
| | |(5060) ------------------> (2829) |
|20.856 | 404 Not here | |SIP
Status
| |(15557) <------------------ (5060) | |
|20.858 | | 200 OK | |SIP
Status
| | |(5060) <------------------ (2829) |
|20.896 | | 487 Request Terminated |SIP
Status
| | |(5060) <------------------ (2829) |
|20.900 | | ACK | |SIP
Request
| | |(5060) ------------------> (2829) |
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