[SR-Users] Problem with forward on busy

[Carsten Bock]

Hi,

do an
- exit;
after t_relay().

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();
>>>  exit; <<<
    }


Otherwise the request get's further processed in the failure_route.

Kind regards,
Carsten

2013/7/25 LAA <ornitorrinco7424 at gmail.com>:
> Excuse me. I have created a new thread by mistake.
>
> ...
>
>
> Hello Hero,
>
> Thanks for your help.
>
> May be I'm loosing something. I have changed my config as you suggested (I
> thing so...):
>
>
>
> 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();
>
>     }
> }
>
> Kamailio sends back 200 OK to the UAC that originated the call, but it never
> sends the new INVITE
>
>
>
>
> |Time     | 192.168.3.20
>
>         | 192.168.0.167                         |
> |         |                   | 192.168.0.197     |
> |3,151    |         INVITE SDP ( telephone-event)          |
> |SIP From: sip:4095 at 192.168.0.197 To:sip:4440 at 192.168.0.197
> |         |(5060)   ------------------>  (5060)   |                   |
> |3,159    |         407 Proxy Authentication Required          |
> |SIP Status
> |         |(5060)   <------------------  (5060)   |                   |
> |3,161    |         ACK       |                   |                   |SIP
> Request
> |         |(5060)   ------------------>  (5060)   |                   |
> |3,161    |         INVITE SDP ( telephone-event)          |
> |SIP From: sip:4095 at 192.168.0.197 To:sip:4440 at 192.168.0.197
> |         |(5060)   ------------------>  (5060)   |                   |
> |3,174    |         100 trying -- your call is important to us          |
> |SIP Status
> |         |(5060)   <------------------  (5060)   |                   |
> |3,174    |                   |         INVITE SDP ( telephone-event)
> |SIP Request
> |         |                   |(5060)   ------------------>  (5060)   |
> |3,176    |                   |         100 Trying|                   |SIP
> Status
> |         |                   |(5060)   <------------------  (5060)   |
> |3,177    |                   |         486 Busy Here                 |SIP
> Status
> |         |                   |(5060)   <------------------  (5060)   |
>
> |3,180    |                   |         ACK       |                   |SIP
> Request
> |         |                   |(5060)   ------------------>  (5060)   |
> |3,195    |         200 OK SDP ( telephone-event)          |
> |SIP Status
> |         |(5060)   <------------------  (5060)   |                   |
> |3,200    |         ACK       |                   |                   |SIP
> Request
> |         |(5060)   ------------------>  (5060)   |                   |
> |3,213    |         RTP (GSM) |                   |                   |RTP
> Num packets:204  Duration:4.069s SSRC:0x8494958
> |         |(49222)  ------------------>  (10028)  |                   |
> |7,288    |         BYE       |                   |                   |SIP
> Request
> |         |(5060)   ------------------>  (5060)   |                   |
> |7,295    |         200 OK    |                   |                   |SIP
> Status
> |         |(5060)   <------------------  (5060)   |                   |
>
>
> what am I loosing?
>
> Regards
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
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http://www.ng-voice.com
mailto:carsten at ng-voice.com

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