23 Feb
2009
23 Feb
'09
2:53 p.m.
Iñaki Baz Castillo wrote:
No puedo lograrlo con telefonos que esten detras de NAT, sucede que al
enviar un INVITE, este se le envia a la IP privada del tel y por supuesto no
llega.
Los telefonos en LAN funcionan bien, y los telefonos haciendo NAT se
registran bien en la BD, mostrando la IP publica y la privada. (en contact
la privada, y en received la publica).
Haciendo una captura con NGREP veo esto, lo cual es logico que esta enviando
el INVITE a la IP privada, no puedo lograr que lo envie a la publica.
IP Asterisk: 200.xx.xx.87
IP Kamailio: 200.xx.xx.53
Tel que llama: 192.168.10.152
Tel que hace NAT: 192.168.2.10
U 192.168.10.152:5060 -> 200.xx.xx.53:5060
INVITE sip:6001@200.xx.xx.53 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.152:5060;branch=z9hG4bK-7d0886bc.
From: "6005" <sip:6005@200.xx.xx.53>;tag=1b7f20bc3b144e87o0.
To: "6001" <sip:6001@200.xx.xx.53>.
Call-ID: 12ee10a0-722d7ff3(a)192.168.10.152.
CSeq: 102 INVITE.
Max-Forwards: 70.
Proxy-Authorization: Digest
username="6005",realm="asterisk",nonce="4883c065",uri="sip:6001@200.xx.xx.53",algorithm=MD5,response="a9
a75f94f05a8e04d2ec7d5ae7ac8def".
Contact: "6005" <sip:6005@192.168.10.152:5060>.
U 200.xx.xx.53:5060 -> 192.168.10.152:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
192.168.10.152:5060;branch=z9hG4bK-7d0886bc;rport=5060;received=192.168.10.152.
From: "6005" <sip:6005@200.xx.xx.53>;tag=1b7f20bc3b144e87o0.
To: "6001" <sip:6001@200.xx.xx.53>.
Call-ID: 12ee10a0-722d7ff3(a)192.168.10.152.
CSeq: 102 INVITE.
Server: Kamailio (1.4.3-notls (i386/linux)).
Content-Length: 0.
U 200.xx.xx.53:5060 -> 200.xx.xx.87:5060
INVITE sip:6001@200.xx.xx.87:5060 SIP/2.0.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=1b7f20bc3b144e87o0>.
Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bKb157.98e72b53.0.
Via: SIP/2.0/UDP 192.168.10.152:5060;rport=5060;branch=z9hG4bK-7d0886bc.
From: "6005" <sip:6005@200.xx.xx.53>;tag=1b7f20bc3b144e87o0.
To: "6001" <sip:6001@200.xx.xx.53>.
Call-ID: 12ee10a0-722d7ff3(a)192.168.10.152.
CSeq: 102 INVITE.
Max-Forwards: 69.
Proxy-Authorization: Digest
username="6005",realm="asterisk",nonce="4883c065",uri="sip:6001@200.xx.xx.53",
U 200.xx.xx.87:5060 -> 200.xx.xx.53:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
200.xx.xx.53;branch=z9hG4bKb157.98e72b53.0;received=200.xx.xx.53.
Via: SIP/2.0/UDP 192.168.10.152:5060;rport=5060;branch=z9hG4bK-7d0886bc.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=1b7f20bc3b144e87o0>.
From: "6005" <sip:6005@200.xx.xx.53>;tag=1b7f20bc3b144e87o0.
To: "6001" <sip:6001@200.xx.xx.53>.
Call-ID: 12ee10a0-722d7ff3(a)192.168.10.152.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:6001@200.xx.xx.87>.
Content-Length: 0.
U 200.xx.xx.87:5060 -> 200.xx.xx.53:5060
INVITE sip:6001@192.168.2.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Contact: <sip:6005@200.xx.xx.87>.
Call-ID: 445adc075a5d751f30d1a306737b80b7(a)200.xx.xx.87.
CSeq: 102 INVITE.
U 200.xx.xx.53:5060 -> 200.xx.xx.87:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport=5060;received=200.xx.xx.87.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Call-ID: 445adc075a5d751f30d1a306737b80b7(a)200.xx.xx.87.
CSeq: 102 INVITE.
Server: Kamailio (1.4.3-notls (i386/linux)).
Content-Length: 0.
U 200.xx.xx.53:5060 -> 192.168.2.10:5060
INVITE sip:6001@192.168.2.10:5060 SIP/2.0.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=as5bbe9873>.
Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bK1604.43fd3526.0.
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport=5060.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Contact: <sip:6005@200.xx.xx.87>.
U 200.xx.xx.87:5060 -> 200.xx.xx.53:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP
200.xx.xx.53;branch=z9hG4bKb157.98e72b53.0;received=200.xx.xx.53.
Via: SIP/2.0/UDP 192.168.10.152:5060;rport=5060;branch=z9hG4bK-7d0886bc.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=1b7f20bc3b144e87o0>.
From: "6005" <sip:6005@200.xx.xx.53>;tag=1b7f20bc3b144e87o0.
To: "6001" <sip:6001@200.xx.xx.53>;tag=as0a184172.
U 200.xx.xx.53:5060 -> 192.168.10.152:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.152:5060;rport=5060;branch=z9hG4bK-7d0886bc.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=1b7f20bc3b144e87o0>.
From: "6005" <sip:6005@200.xx.xx.53>;tag=1b7f20bc3b144e87o0.
To: "6001" <sip:6001@200.xx.xx.53>;tag=as0a184172.
Call-ID: 12ee10a0-722d7ff3(a)192.168.10.152.
U 200.xx.xx.53:5060 -> 192.168.2.10:5060
INVITE sip:6001@192.168.2.10:5060 SIP/2.0.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=as5bbe9873>.
Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bK1604.43fd3526.0.
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport=5060.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Contact: <sip:6005@200.xx.xx.87>.
Call-ID: 445adc075a5d751f30d1a306737b80b7(a)200.xx.xx.87.
U 200.xx.xx.53:5060 -> 192.168.2.10:5060
INVITE sip:6001@192.168.2.10:5060 SIP/2.0.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=as5bbe9873>.
Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bK1604.43fd3526.0.
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport=5060.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Contact: <sip:6005@200.xx.xx.87>.
Call-ID: 445adc075a5d751f30d1a306737b80b7(a)200.xx.xx.87.
U 200.xx.xx.53:5060 -> 192.168.2.10:5060
INVITE sip:6001@192.168.2.10:5060 SIP/2.0.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=as5bbe9873>.
Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bK1604.43fd3526.0.
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport=5060.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Contact: <sip:6005@200.xx.xx.87>.
Call-ID: 445adc075a5d751f30d1a306737b80b7(a)200.xx.xx.87.
U 200.xx.xx.53:5060 -> 192.168.2.10:5060
INVITE sip:6001@192.168.2.10:5060 SIP/2.0.
Record-Route: <sip:200.xx.xx.53;lr=on;ftag=as5bbe9873>.
Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bK1604.43fd3526.0.
Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK0105fa70;rport=5060.
From: "6005" <sip:6005@200.xx.xx.87>;tag=as5bbe9873.
To: <sip:6001@192.168.2.10:5060>.
Contact: <sip:6005@200.xx.xx.87>.
Call-ID: 445adc075a5d751f30d1a306737b80b7(a)200.xx.xx.87.
La configuracion de mi CFG es:
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
route(2);
if (has_totag()) {
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(1);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
if ($rU =~ "5[0-9]" && src_ip!=200.xx.xx.87){
route(3);
exit;
}
setflag(1); # do accounting
}
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
}
# requests for my domain
if (is_method("PUBLISH"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER"))
{
if (isflagset(5)) {
setbflag(6);
save("location");
};
if (!save("location"))
sl_reply_error();
append_hf("P-hint: usrloc applied\r\n");
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
if (!lookup("location")) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[3] {
log(1, "Reenvia a Asterisk \n");
rewritehostport("200.xx.xx.87:5060");
route(1);
}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])")
{
force_rtp_proxy();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
--
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Mi idea es que los telefonos puedas comunicarse
entre si sin llegar a
Asterisk, pero si desean conferencias, voicemail, etc, la llamada se
efectue al asterisk.
Perfectamente viable. Hay ejemplos de ello en voip-info y en el wiki de
Kamailio.