I have checked that I'm , experiencing the same problem when the redirection to voicemail is originated by the destination UAC via 302 message. Kamailio sends the packet to the destination UAC, even when I set $du to null. ??¿?¿?¿¿??¿
if ($rU=~"^voicemail.*") {
$du = $null; remove_hf("P-App-Name"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: mod=box;usr=$rU;dom=sipproxy.a.com ;uid=$rU;did=sipproxy.a.com;\r\n"); $ru = "sip:" + $rU + "@" + "192.168.0.197:5080"; $du = $null; route(RELAY); exit; }
Conv.| Time | 192.168.3.20 | 192.168.0.167 | | | | 192.168.0.197 | 0 |3,574 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | 0 |3,575 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | 0 |3,577 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | 0 |3,577 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | 0 |3,584 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | 0 |3,585 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | 0 |3,587 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | 0 |3,587 | | 302 Moved Temporarily |SIP Status | | |(5060) <------------------ (5060) | 0 |3,588 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | 0 |3,592 | 302 Moved Temporarily | |SIP Status | |(5060) <------------------ (5060) | | 0 |3,594 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | ----------------------------------------------------------------------------- 1 |3,596 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:voicemail4440@192.168.0.167:5060 | |(5060) ------------------> (5060) | | 1 |3,596 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | 1 |3,600 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | 1 |3,601 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:voicemail4440@192.168.0.167:5060 | |(5060) ------------------> (5060) | | 1 |3,608 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | 1 |3,608 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | 1 |3,608 | | 404 Not Found |SIP Status | | |(5060) <------------------ (5060) | 1 |3,609 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | 1 |3,614 | 404 Not Found | |SIP Status | |(5060) <------------------ (5060) | | 1 |3,615 | ACK | | |SIP Request | |(5060) ------------------> (5060) | |
2013/7/25 LAA ornitorrinco7424@gmail.com
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 = $null; #append_branch(); route(RELAY); #t_relay();
}}
And kamailio gets into a strange behavior
|Time | 192.168.3.20 | 192.168.0.167 | | | | 192.168.0.197 | |3,366 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |3,370 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | |
|3,380 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,382 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |3,393 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |3,394 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |3,395 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |3,395 | | 486 Busy Here |SIP Status | | |(5060) <------------------ (5060) | |3,398 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |3,416 | 500 I'm terribly sorry, server error occurred ...SL) | |SIP Status | |(5060) <------------------ (5060) | | |3,416 | 486 Busy Here | |SIP Status | |(5060) <------------------ (5060) | | |3,418 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,418 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,872 | 486 Busy Here | |SIP Status | |(5060) <------------------ (5060) | | |3,873 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |4,875 | 486 Busy Here | |SIP Status | |(5060) <------------------ (5060) | | |4,876 | ACK | | |SIP Request | |(5060) ------------------> (5060) | |
Are you using this sequence within failure route? or in the call routing section? I'm using this sequence in the route section that is working OK:
if ($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@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@voicemail:5060" and then just do the normal relay route to deliver the message. Your *_hf shouldn't have any effect on routing.
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