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.
2013/7/25 Daniel Tryba <dan...@pocos.nl>
> On Thursday 25 July 2013 16:30:21 you wrote:
>
> > 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 = $null;
> > #$du = "sip:192.168.0.197";
> > #append_branch();
> > t_relay();
>
> Taking a look at my config which I found to work after the long struggle
> you
> are experiencing right now.
>
> if($avp(dst_voicemail))
> {
> $du=$null;
> $ru = "sip:tovm-" + $avp(dst_voicemail) + "@" +
> $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port);
> route(RELAY);
>
> exit;
> }
>
> Which effectively sets $du to null (if not null the message would get
> relayed
> to the original destination (the proxy itself)) and rewrites $ru to
> something
> like
> "sip:tovm-0123456789@voicemail:5060"
> and then just do the normal relay route to deliver the message. Your *_hf
> shouldn't have any effect on routing.
>
> --
>
> POCOS B.V. - Croy 9c - 5653 LC Eindhoven
> Telefoon: 040 293 8661 - Fax: 040 293 8658
> http://www.pocos.nl/ - http://www.sipo.nl/
> K.v.K. Eindhoven 17097024
>
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