Can anyone help me here.
--- On Fri, 2/19/10, Slot Zero slotzero1@yahoo.com wrote:
From: Slot Zero slotzero1@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[0-9]@*") {
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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