Kamailio 3.0.0: Problemas con Forward on Busy - sr-users-es

22 Jul 2013


      Hola a todos,
Estoy tratando de configurar Kamailio 3.0.0 para que redirija las  llamadas
a un buzón de voz basado en SEMS cuando el usuario llamado está  ocupado.
Siguiendo las instrucciones de  varios ejemplos encontrados en internet, y
del propio script ejemplo de kamailio.cfg, estoy usando las funciones
revert_uri(), rewritehostport() (o reescribiéndo el R-URI manualmente) y
append_branch(), dentro de la failure_route. Aparentemente modifica
correctamente el R-URI y crea la segunda "pata" de la llamada. Sin embargo,
a pesar de que la dirección del R-Uri aparece bien cambiada como decía en
la Request Line, el paquete IP se envía a la dirección IP del destinatario
original:
|Time     | 192.168.3.20
        | 192.168.0.167                         |
|         |                   | 192.168.0.197     |
|5,069    |         INVITE SDP ( telephone-event)
|                   |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
|         |(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@192.168.0.197
To:sip:4440@192.168.0.197
|         |(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)   |                   |
Esta es la captura del Invite de la redirección, en el 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@192.168.0.197:5080, 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@192.168.0.197:5080 SIP/2.0
        Method: INVITE
        Request-URI: sip:voicemail4440@192.168.0.197:5080
        [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@192.168.3.20:5060
        Call-ID: 8EAF9EC2-1DD2-11B2-B110-C84E476664B0@10.0.3.15
        Content-Type: application/sdp
        CSeq: 2 INVITE
        From: "4095"sip:4095@192.168.0.197;tag=121754238352072516
        Max-Forwards: 69
        To: sip:4440@192.168.0.197
        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
La configuración que he usado es la siguiente:
# 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@192.168.0.102");
        append_branch();
        # do not set the missed call flag again
        t_relay();
    }
}
¿Alguien ha tenido un problema similar? Cualquier ayuda será bienvenida.
Saludos.
LAA.