Re: [Kamailio-Users-ES] Modificar Headers

23 Feb 2009


      Iñaki Baz Castillo wrote:
...
...
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.
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@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@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@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@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@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@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@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@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@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@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@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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Re: [Kamailio-Users-ES] Modificar Headers