Can anyone help me here.
--- On Fri, 2/19/10, Slot Zero <slotze...@yahoo.com> wrote:
> From: Slot Zero <slotze...@yahoo.com>
> Subject: [Kamailio-Users] Forward calls from Asterisk to SIP provider via
> Kamailio for termination
> To: users@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...@*")
> {
>
> 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);
> }
>
>
>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users@lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
