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 <ornitorrinco7...@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@192.168.0.197 To:sip:4440@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@192.168.0.197 To:sip:4440@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@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users