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*>
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