[SR-Users] Problem with forward on busy

[LAA]

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 at 192.168.0.197
To:sip:4440 at 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 at 192.168.0.197
To:sip:4440 at 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 <daniel at 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 at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20130725/b54915f7/attachment.html>