[Serusers] Need help or i give up

Sanjay Duggal Sanjayd at Pressis.com
Fri Jun 18 06:05:55 CEST 2004


Hi All

    I have been working to make this work.
    I have an Asterisk gateway and a Ser proxy running on to different
servers.
    Ser has one Public Ip and a private one. ( on two different NICs)
    I have installed RTPProxy on the same server as Ser.
    When I make a call from my cell phone I can have a conversion on my SIP
    phone. Everything works great.
    But when I call from SIP phone it is no sound.
    My cell rings I can pick it up but no sound.
    Can one of you expert on this mater please help me?
    I'm enclosing my ser.cfg 
    #
    # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
    #
    # simple quick-start config script
    #
    # ----------- global configuration parameters ------------------------

    
    debug=4         # debug level (cmd line: -dddddddddd)
    	fork=no
    	log_stderror=yes	# (cmd line: -E)
    
    /* Uncomment these lines to enter debugging mode 
     debug=9
     fork=yes
     log_stderror=yes
    */
    
    	alias="Domain.com"
    	alias="my.Domain.com"
    # 	alias="192.168.0.100"
    # 	alias="192.168.0.200"
    
    
    	listen="public ip" 
    	check_via=no	# (cmd. line: -v)
    	dns=no           # (cmd. line: -r)
    	rev_dns=no      # (cmd. line: -R)
    #port=5060
    #children=4
    	fifo="/tmp/ser_fifo"
    	fifo_mode=0777
    
    	# ------------------ module loading
    ----------------------------------
    
    # Uncomment this if you want to use SQL database
    	loadmodule "/usr/local/lib/ser/modules/mysql.so"
    
    	loadmodule "/usr/local/lib/ser/modules/sl.so"
    	loadmodule "/usr/local/lib/ser/modules/tm.so"
    	loadmodule "/usr/local/lib/ser/modules/rr.so"
    	loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
    	loadmodule "/usr/local/lib/ser/modules/usrloc.so"
    	loadmodule "/usr/local/lib/ser/modules/registrar.so"
    	loadmodule "/usr/local/lib/ser/modules/textops.so"
    
    	loadmodule "/usr/local/lib/ser/modules/nathelper.so"
    
    # Uncomment this if you want digest authentication
    # mysql.so must be loaded !
    	loadmodule "/usr/local/lib/ser/modules/auth.so"
    	loadmodule "/usr/local/lib/ser/modules/auth_db.so"
    
    	# ----------------- setting module-specific parameters
    ---------------
    	# -- usrloc params --
    
    	modparam("usrloc", "db_mode",   0)
    
    # Uncomment this if you want to use SQL database 
    # for persistent storage and comment the previous line
    	modparam("usrloc", "db_mode", 2)
    
    	# -- auth params --
    # Uncomment if you are using auth module
    	#
    	modparam("auth_db", "calculate_ha1", yes)
    	#
    # If you set "calculate_ha1" parameter to yes (which true in this
config), 
    # uncomment also the following parameter)
    	#
    	modparam("auth_db", "password_column", "password")
    
    	# -- rr params --
    # add value to ;lr param to make some broken UAs happy
    	modparam("rr", "enable_full_lr", 1)
    
    	# !! Nathelper
    	modparam("registrar", "nat_flag", 6)
    	modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
    # modparam("nathelper", "ping_nated_only", 1)   # Ping only clients
behind
    NAT
    	# -------------------------  request routing logic
    -------------------
    
    # main routing logic
     
    	route {
    # initial sanity checks -- messages with
    # max_forwards==0, or excessively long requests
            if (!mf_process_maxfwd_header("10")) {
                    sl_send_reply("483","Too Many Hops");
                    break;
            };
            if (msg:len >=  max_len ) {
                    sl_send_reply("513", "Message too big");
                    break;
            };
     
            # !! Nathelper
    # Special handling for NATed clients; first, NAT test is
    # executed: it looks for via!=received and RFC1918 addresses
    # in Contact (may fail if line-folding is used); also,
    # the received test should, if completed, should check all
    # vias for rpesence of received
            if (nat_uac_test("3")) {
    		log(1, "NAT client\n");
    	record_route();	
    
    # Allow RR-ed requests, as these may indicate that
    # a NAT-enabled proxy takes care of it; unless it is
    # a REGISTER
    #if (method=="REGISTER") {
    
       		if (method == "REGISTER" || !search("^Record-Route:")) {
                    	log(1, "LOG: Someone trying to register from private
    IP, rewriting\n");
     
    # This will work only for user agents that supportsymmetric
    # communication. We tested quite many of them andmajority is
    # smart enough to be symmetric. In some phones it takesa configuration
    # option. With Cisco 7960, it is called NAT_Enable=Yes,with kphone it is
    # called "symmetric media" and "symmetric signalling".
    
                        	fix_nated_contact(); # Rewrite contact with
source
    IPof signalling
    
    			if (method == "INVITE") {
    				log(1, "NAT -> INVITE\n");
    				fix_nated_sdp("1"); # Add direction=active
    to SDP
    			};
    	        
    			force_rport(); # Add rport parameter to topmost Via
                  		setflag(6);    # Mark as NATed
    			append_to_reply("P-NATed-Caller: Yes\r\n");
                    };
    	};
    # we record-route all messages -- to make sure that
    # subsequent messages will go through our proxy; that's
    # particularly good if upstream and downstream entities
    # use different transport protocol
    	if (!method=="REGISTER")
    		record_route();
     
    # subsequent messages withing a dialog should take the
    # path determined by record-routing
    	if (loose_route()) {
    # mark routing logic in request
    		append_hf("P-hint: rr-enforced\r\n");
    		route(1);
    		break;
    	};
    
    	if (uri=~"^sip:0[0-9]*@my.Domain.com") {
    		log(1, "Forwarding to Asteriks\n");
    		route(1);
    #		rewritehostport("192.168.0.200:5060");
    #		append_hf("P-hint: GATEWAY\r\n");
    #		t_relay_to_udp("192.168.0.200", "5060");
    		#forward(192.168.0.200,5060);
                    #      Where local asterisk is listening
    		#t_relay();
    		break;
    	};
    	if (!uri==myself) {
    # mark routing logic in request
    		append_hf("P-hint: outbound\r\n");
    		route(1);
    		break;
    	};
     
    # if the request is for other domain use UsrLoc
    	# (in case, it does not work, use the following command
    # with proper names and addresses in it)
    	if (uri==myself) {
     
    		if (method=="REGISTER") {
    			log(1, "Myself -> REGISTER\n");
    # Uncomment this if you want to use digest authentication
    #                        if (!www_authorize("iptel.org", "subscriber"))
{
    #                                www_challenge("iptel.org", "0");
    #                                break;
    			#                        };
     
    			save("location");
    			break;
    		};
     
    		lookup("aliases");
    		if (!uri==myself) {
    			append_hf("P-hint: outbound alias\r\n");
    route(1);
    			break;
    		};
     
    # native SIP destinations are handled using our USRLOC DB
    		if (!lookup("location")) {
    			sl_send_reply("404", "Not Found");
    			break;
    		};
    	};
    
    #inserted by klaus
    	if (method=="INVITE") {
    		log(1, "INVITE\n");
    		record_route();
    		force_rtp_proxy();
    		/* set up reply processing */
    		t_on_reply("1");
    	};
    
    
    
    # forward to current uri now; use stateful forwarding; that
    # works reliably even if we forward from TCP to UDP
    	if (!t_relay()) {
    		sl_reply_error();
    	};
    
    
    
    #  append_hf("P-hint: usrloc applied\r\n");        route(1);
    }
     
    	route[1] {
    		# !! Nathelper
    		log(1, "ROUTE[1]\n");
    		if
    (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
    search("^Route:")){
    			sl_send_reply("479", "We don't forward to private
    IPaddresses");
    			break;
    		};
            
                    # if client or server know to be behind a NAT, enable
relay
    		if (isflagset(6)) {
    			log(1, "Flag is 6 (NAT)\n");
    			if (!is_present_hf("P-RTP-Proxy")) {
    				force_rtp_proxy();
    				append_hf("P-RTP-Proxy: YES\r\n");
    			};
    			append_hf("P-NATed-Calee: Yes\r\n");
    
    		rewritehostport("192.168.0.200:5060");
                   append_hf("P-hint: GATEWAY\r\n");
                   t_relay_to_udp("192.168.0.200", "5060");
    		break ;
    		};
    		
    
    	
     
    # NAT processing of replies; apply to all transactions (forexample,
    # re-INVITEs from public to private UA are hard to identify as
    # NATed at the moment of request processing); look at replies
    		t_on_reply("1");
     
    # send it out now; use stateful forwarding as it works reliably
    # even for UDP2TCP
    		if (!t_relay()) {
    			log(1, "ROUTE[1] -> sl_reply_error\n");
    			sl_reply_error();
    			break;
    		};
    
    	}
     
    # !! Nathelper
    onreply_route[1] {
    	log(1, "onreply_route[1]\n");
    	if (status=~"[12][0-9][0-9]"){
    	 
     #               force_rtp_proxy();
    
    
    #	if (isflagset(6) && status=~"(183)|2[0-9][0-9]") {
    		fix_nated_contact();
    		fix_nated_sdp("1");
    		force_rtp_proxy();
    	} else #if(nat_uac_test("1"))
    		{
    			fix_nated_contact();
    			force_rtp_proxy();
    		};
     
    }
      
    
    
    
    regards 
    
    ------------------------------------------------------------- 
    Pressis Consulting DA                   
    Sanjay Duggal                           
    Operative CTO                           
    
 

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