25 Jul
2013
25 Jul
'13
5:51 p.m.
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