[Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio for termination

Slot Zero slotzero1 at yahoo.com
Mon Feb 22 13:06:26 CET 2010


Can anyone help me here.

--- On Fri, 2/19/10, Slot Zero <slotzero1 at yahoo.com> wrote:

> From: Slot Zero <slotzero1 at yahoo.com>
> Subject: [Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio for termination
> To: users at lists.kamailio.org
> Date: Friday, February 19, 2010, 7:10 PM
> Hello,
> 
> I am a Kamailio noob :). I am trying to get Asterisk to
> forward calls to 
> my SIP provider via Kamailio. The same machine is running
> Kamailio and 
> Asterisk. I do not want to consume credentials as they have
> to be passed 
> on all the way to my SIP provider. There is no NAT of any
> sorts. SIP 
> Phone/Users connect to Asterisk.I do not need to
> authenticate when forwarding call from Asterisk to Kamailio
> as they are 
> both running on the same server but I do need to make sure
> that Kamailio 
> dials and forwards 011+number to be sent from local host
> port 
> 5062(Asterisk listener) to SIP provider only. I have 6
> Public IP addresses 
> mapped on the server. I want to use the force_send_socket
> to allow me to 
> change source IP of SIP requests when being sent to the SIP
> provider on the basis of credentials username in the
> request. I have pasted my config below. Please tell me what
> am I doing wrong here. In the kamctlrc file i have
> SIP_DOMAIN=localhost
> 
> Thank you
> 
> #------CONFIG BEGINS------------------
> mpath="/lib/kamailio/modules_k/"
> 
> debug=3
> fork=yes
> 
> children=4
> auto_aliases=no
> alias=localhost
> alias=192.168.10.1
> alias=192.168.10.2
> alias=192.168.10.3
> alias=192.168.10.4
> alias=192.168.10.5
> alias=192.168.10.6
> 
> disable_tcp=yes
> 
> loadmodule "sl.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "/lib/kamailio/modules/tm.so"
> loadmodule "textops.so"
> 
> modparam("rr", "enable_full_lr", 1)
> 
> route {
>         # Sanity Check
>         # ------------
> 
>         # filter too old messages
>         if
> (!mf_process_maxfwd_header("10")) {
>                
> log("LOG: Too many hops\n");
>                
> sl_send_reply("483","Too Many Hops");
>                
> break;
>         };
> 
> 
>         if(msg:len>2048) {
>            
>    sl_send_reply("413", "message too large to
> be forwarded over UDP without fragmentation");
>            
>    exit;
>         };
> 
>         # Record Route and NAT Preset
>         # --------------------
>         if (method != "REGISTER") {
>                
> record_route();
>         };
> 
>         # Loose Route
> 
>         # -----------
>         if (loose_route()) {
> 
>                
> route(1);
>                
> return;
>         };
> 
>         # Call Type Processing
>         # --------------------
>         if (uri != myself) {
>                
> route(1);
>                
> return;
>         };
> 
>         if (uri == myself) {
>                 if
> (method == "BYE") {
>                
>         route(4);
>                
>         return;
>                 }
> else if (method == "CANCEL") {
>                
>         route(4);
>                
>         return;
>                 }
> else if (method == "INVITE") {
>                
>         route(3);
>                
>         return;
>                 }
> else if (method == "NOTIFY") {
>                
>         sl_send_reply("200",
> "Understood");
>                
>         return;
>                 }
> else if (method == "OPTIONS") {
>                
>         sl_send_reply("200", "Got it");
>                
>         return;
>                 }
>         };
>         route(1);
> }
> 
> # Default Message Handling
> # -----------------------
> route[1] {
>        t_on_reply("1");
>         if (!t_relay()) {
>                
> sl_reply_error();
>         };
> }
> 
> # INVITE Message Handling
> # ----------------------------------
> 
> # ----------------------------------
> route[3] {
>         if (uri =~ "^sip:011[0-9]@*")
> {
>                
> rewritehostport("sip.voipprovider.com:5060");
>                 if
> (search("^(Contact|m): .*user01*@(127\.0\.0\.1|localhost)"))
> {
>                
> force_send_socket(192.168.0.2:5060);
>                 };
>                
> route(1);
>                
> return;
>         };
> 
> }
> 
> # CANCEL and BYE Message Handling
> # ----------------------------------
> route[4] {
>         route(1);
> }
> 
> 
> 
>       
> 
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> 


      



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