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
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@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