OK, Daniel and thanks for your help,
I see that you don't append brach but you are calling route(RELAY) instead of t_relay() directly. I have tryed with this configuration within failure route:
if (t_check_status("486|408")) {
#revert_uri();
prefix("voicemail");
remove_hf("P-App-Name");
append_hf("P-App-Name: voicemail\r\n");
append_hf("P-App-Param: mod=box;usr= $rU;dom=
sipproxy.a.com;uid=$rU;did=
sipproxy.a.com;\r\n");
rewritehostport("
192.168.0.197:5080");
$du = $null;
#append_branch();
route(RELAY);
#t_relay();
}
}
And kamailio gets into a strange behavior
|Time | 192.168.3.20 | 192.168.0.167 |
| | | 192.168.0.197 |
|3,366 | INVITE SDP ( telephone-event) | |SIP From:
sip:4095@192.168.0.197 To:sip:4440@192.168.0.197
| |(5060) ------------------> (5060) | |
|3,370 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|3,380 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|3,382 | INVITE SDP ( telephone-event) | |SIP From:
sip:4095@192.168.0.197 To:sip:4440@192.168.0.197
| |(5060) ------------------> (5060) | |
|3,393 | 100 trying -- your call is important to us | |SIP Status
| |(5060) <------------------ (5060) | |
|3,394 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|3,395 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5060) |
|3,395 | | 486 Busy Here |SIP Status
| | |(5060) <------------------ (5060) |
|3,398 | | ACK | |SIP Request
| | |(5060) ------------------> (5060) |
|3,416 | 500 I'm terribly sorry, server error occurred ...SL) | |SIP Status
| |(5060) <------------------ (5060) | |
|3,416 | 486 Busy Here | |SIP Status
| |(5060) <------------------ (5060) | |
|3,418 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|3,418 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|3,872 | 486 Busy Here | |SIP Status
| |(5060) <------------------ (5060) | |
|3,873 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|4,875 | 486 Busy Here | |SIP Status
| |(5060) <------------------ (5060) | |
|4,876 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
Are you using this sequence within failure route? or in the call routing section? I'm using this sequence in the route section that is working OK:
if ($rU=~"^voicemail.*") {
remove_hf("P-App-Name");
append_hf("P-App-Name: voicemail\r\n");
append_hf("P-App-Param: mod=box;usr=$rU;dom=
sipproxy.a.com;uid=$rU;did=
sipproxy.a.com;\r\n");
$ru = "sip:" + $rU + "@" + "
192.168.0.197:5080";
route(RELAY);
exit;
}
The problem is when I try to get a call forwarded by kamailio to voice mail when it gets a busy message to the destination message. In your implementation are you expecting a 302 (temporary unavailable) message from the destination UAC?
Regards.
L.