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
LAA
*************
had the same issue here. you have to manually set $du=$null, else it
doesn't get reset for the failure branch.
On 7/23/13, LAA <ornitorrinco7424 at
gmail.com
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>
wrote:
* Hi all,*>**>* I'm running Kamailio 3.0.0,
with SEMS integration as Media Server for Voice*>* mail. I'm trying to get a
configuration to forward calls on busy to voice*>* mail. I have followed without
success some examples. I'm using*>* revert_uri(), rewritehostport() and
append_branch(), within failure_route.*>* It seems to be modifying R-URI properly, and
generating the new branch, but*>* Kamailio is sending the new invite packet to the IP
address of the original*>* destination UAC, and not to the IP address of the voicemail,
that was*>* indicated in the R-URI. Here you can see the packet flow:*>**>* |Time
| 192.168.3.20*>* | 192.168.0.167 |*>* |
| | 192.168.0.197 |*>* |5,069 | INVITE SDP (
telephone-event)*>* | |SIP From: sip:4095 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* To:sip:4440 at
192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>*
| |(5060) ------------------> (5060) | |*>* |5,071
| 407 Proxy Authentication Required*>* | |SIP Status*>* |
|(5060) <------------------ (5060) | |*>* |5,074 |
ACK | | |SIP*>* Request*>* |
|(5060) ------------------> (5060) | |*>* |5,076 |
INVITE SDP ( telephone-event)*>* | |SIP From: sip:4095 at
192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>*
To:sip:4440 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* |
|(5060) ------------------> (5060) | |*>* |5,084 |
100 trying -- your call is important to us*>* | |SIP Status*>* |
|(5060) <------------------ (5060) | |*>* |5,085 |
| INVITE SDP (*>* telephone-event) |SIP Request*>*
| | |(5060) ------------------> (5060) |*>* |5,088
| | 100 Trying| |SIP*>* Status*>* |
| |(5060) <------------------ (5060) |*>* |5,088 |
| 486 Busy Here |SIP*>* Status*>* | |
|(5060) <------------------ (5060) |*>* |5,091 |
| ACK | |SIP*>* Request*>* | |
|(5060) ------------------> (5060) |*>* |5,101 |
| INVITE SDP (*>* telephone-event) |SIP Request*>* | |
|(5060) ------------------> (5060) |*>* |5,102 |
| 404 Not Found |SIP*>* Status*>* | |
|(5060) <------------------ (5060) |*>* |5,102 | |
ACK | |SIP*>* Request*>* | |
|(5060) ------------------> (5060) |*>* |5,103 | 404 Not Found
| |SIP*>* Status*>* | |(5060)
<------------------ (5060) | |*>* |5,106 | ACK
| | |SIP*>* Request*>* | |(5060)
------------------> (5060) | |*>**>* And the RAW capture of
the INVITE message in timestamp 5,101.*>**>**>**>* No. Time Source
Destination Protocol*>* Info*>* 1235 5.100698
192.168.0.197 192.168.0.167 SIP/SDP*>* Request: INVITE
sip:voicemail4440 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5080, with
session*>* description*>**>* Frame 1235 (1151 bytes on wire, 1151 bytes
captured)*>* Ethernet II, Src: CadmusCo_96:31:84 (08:00:27:96:31:84), Dst:*>*
Micro-St_6d:77:54 (00:21:85:6d:77:54)*>* Internet Protocol, Src: 192.168.0.197
(192.168.0.197), Dst: 192.168.0.167*>* (192.168.0.167)*>* User Datagram Protocol,
Src Port: sip (5060), Dst Port: sip (5060)*>* Session Initiation Protocol*>*
Request-Line: INVITE sip:voicemail4440 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5080 SIP/2.0*>*
Method: INVITE*>* Request-URI: sip:voicemail4440 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5080*>*
[Resent Packet: True]*>* [Suspected resend of frame: 1233]*>* Message
Header*>* Record-Route: <sip:192.168.0.197;lr=on;nat=*>* yes>*>*
Via: SIP/2.0/UDP 192.168.0.197;branch=z9hG4bKafce.403718a6.1*>* Via:
SIP/2.0/UDP*>*
192.168.57.20;received=192.168.3.20;rport=5060;branch=z9hG4bK0a00030f0000003151ed60b85ec2c3de000000c8*>*
Content-Length: 386*>* Contact: <sip:4095 at 192.168.3.20
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060>*>*
Call-ID: 8EAF9EC2-1DD2-11B2-B110-C84E476664B0 at 10.0.3.15
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>*
Content-Type: application/sdp*>* CSeq: 2 INVITE*>* From:
"4095"<sip:4095 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>;tag=121754238352072516*>*
Max-Forwards: 69*>* To: <sip:4440 at 192.168.0.197
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>*>*
User-Agent: SJphone/1.60.299a/L (SJ Labs)*>* P-App-Name: voicemail*>*
P-App-Param: mod=box;usr= voicemail4440;dom=sipproxy.a.com*>*
;uid=voicemail4440;did=sipproxy.a.com;*>* Message Body*>**>* Here you can see
the failure_route in my kamailio.cfg file:*>**>* # Sample failure route*>*
failure_route[FAIL_ONE] {*>* #ifdef WITH_NAT*>* if
(is_method("INVITE")*>* && (isbflagset("6") ||
isflagset(5))) {*>* unforce_rtp_proxy();*>* }*>* #endif*>**>*
if (t_is_canceled()) {*>* exit;*>* }*>**>* # uncomment the
following lines if you want to block client*>* # redirect based on 3xx
replies.*>* ##if (t_check_status("3[0-9][0-9]")*>* ) {*>*
##t_reply("404","Not found");*>* ## exit;*>*
##}*>**>* # uncomment the following lines if you want to redirect the
failed*>* # calls to a different new destination*>* 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");*>* $ru = "sip:" + $rU +
"@" + "192.168.0.197:5080";*>*
#rewritehostport("192.168.0.197:5080");*>*
#append_branch("sip:4888 at 192.168.0.102
<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>");*>*
append_branch();*>* # do not set the missed call flag again*>*
t_relay();*>* }*>* }*>**>* Has anybody experienced this problem? Any help
would be wellcome*>**>* Best Regards*>**>* LAA*>