Here is my ser.cg file. I try to configure send IM, Missed calls,
Voicemail.
I just can configure Send IM.
Now I can call isdngw, ivr,..
I need help to configure lookup location logic for missed calls and
voicemail (offline users)
Anybody could help me to configure the others functions (problem with
logic !!).
harry
Regards
#
# $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=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=192.168.0.1
port=5060
children=4
fifo="/tmp/ser_fifo"
fifo_mode=0666
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/domain.so"
loadmodule "/usr/lib/ser/modules/msilo.so"
loadmodule "/usr/lib/ser/modules/acc.so"
loadmodule "/usr/lib/ser/modules/vm.so"
loadmodule "/usr/lib/ser/modules/uri.so"
loadmodule "/usr/lib/ser/modules/group.so"
loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/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_url", "mysql://ser:heslo@localhost/ser")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", 1)
# -- 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)
# --registrar params--
modparam("registrar", "use_domain", 1)
# --domain params--
modparam("domain", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("domain", "db_mode", 1) # Use chaching
# --acc params--
modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("acc", "db_missed_flag", 3)
# --msilo params--
modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("msilo", "db_table", "silo")
modparam("msilo","registrar","sip:registrar@192.168.0.1")
# --uri params--
#modparam("uri", "db_url", "mysql://ser:heslo@localhost/ser")
#modparam("uri", "subscriber_table", "subscriber")
# --vm params--
modparam("voicemail", "db_url", "mysql://ser:heslo@localhost/ser")
# ------------------------- 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;
};
# 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
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# --isdn gateway--
if(method=="INVITE" || method=="BYE" || method=="CANCEL"){
if (uri=~"sip:[0-9]{10}@.*") {
route(2);
break;
};
};
# --ivr conf--
if(method=="INVITE" || method=="BYE" || method=="CANCEL"){
if (uri=~"sip:5000@.*") {
route(4);
break;
};
};
# --conference--
if(method=="INVITE" || method=="BYE" || method=="CANCEL"){
if (uri=~"sip:6000@.*") {
route(5);
break;
};
};
# --play an annoucement--
if(method=="INVITE" || method=="BYE" || method=="CANCEL"){
if (uri=~"sip:7000@.*") {
route(6);
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 (is_from_local()) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
break;
};
save("location");
# MSILO - dumping user's offline messages
if (m_dump())
{
log("MSILO: offline messages dumped - if they were\n");
}else{
log("MSILO: no offline messages dumped\n");
};
break;
};
# does the user wish redirection on no availability? (i.e., is he
# in the voicemail (ser->grp) group?)
if (is_user_in("Request-URI", "voicemail")) {
t_on_failure("4");
setflag(4);
};
# native SIP destinations are handled using our USRLOC DB
lookup("aliases");
if (!lookup("location")) {
if (! t_newtran()) {
sl_reply_error();
break;
};
# we do not care about anything else but MESSAGEs
if (!method=="MESSAGE") {
if (!t_reply("404", "Not found")) {
sl_reply_error();
};
break;
};
log("MESSAGE received -> storing using MSILO\n");
# MSILO - storing as offline message
if (m_store("0")) {
log("MSILO: offline message stored\n");
if (!t_reply("202", "Accepted")) {
sl_reply_error();
};
}else{
log("MSILO: offline message NOT stored\n");
if (!t_reply("503", "Service Unavailable")) {
sl_reply_error();
};
};
break;
};
# if the downstream UA does not support MESSAGE requests
# go to failure_route[1]
t_on_failure("1");
t_relay();
break;
};
# 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();
};
}
failure_route[1] {
# forwarding failed -- check if the request was a MESSAGE
if (!method=="MESSAGE")
{
break;
};
log(1,"MSILO: the downstream UA does not support MESSAGE requests ...\n");
# we have changed the R-URI with the contact address -- ignore it now
if (m_store("1"))
{
log("MSILO: offline message stored\n");
t_reply("202", "Accepted for delivery");
}else{
log("MSILO: offline message NOT stored\n");
t_reply("503", "Service Unavailable");
};
}
failure_route[4] {
route(3);
#append_branch("sip:80000@10.1.2.5");
append_urihf("CC-Diversion: ", "\r\n");
append_hf("P-hint: OFFLINE-VOICEMAIL\r\n");
t_relay();
}
route[2] {
# ############################## #
# isdngw specific configuration #
# ############################## #
if(t_newtran()){
if(method=="INVITE" || method=="BYE" || method=="CANCEL"){
# send a response right at the start to avoid retransmissions
t_reply("100","Trying -- just wait a minute !");
# isdngw only gets activated on invite requests
if(method=="INVITE"){
# filename is defined in sems.conf.
if(uri=~"sip:[0-9]{10}@.*"){
if(!vm("/tmp/am_fifo","isdngw")){
log("could not contact isdngw\n");
t_reply("500","could not contact isdngw");
};
# Allow the announcement module of sems to be used as well.
# This can be useful for testing the isdngw.
} else if(uri=~"sip:7000@.*"){
if(!vm("/tmp/am_fifo","announcement")){
log("could not contact announcement\n");
t_reply("500","could not contact announcement");
};
# we dont feel responsible for sip addresses of any other kind,
# so send the right error code.
} else {
t_reply("404","Not Found");
};
# stop routing here, the message is now processed by the media server
break;
};
# The following handles the call termination, we must pass these requests
# to the media server as follows. Again make shure the fifo name and permissions
# are set correctly (like im sems.conf).
if((method=="BYE")||(method=="CANCEL")){
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the media server\n");
t_reply("500","could not contact the media server");
};
break;
};
# other methods than INVITE, BYE and CANCEL are not handled by this SIP Server
# so we sent an error message
} else {
log("ERROR: method not supported\n");
t_reply("500", "sorry, method not supported");
};
} else {
# for any reason the transaction could not be created, send error code
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
};
# in isdngw.conf. Don't change this setting.
t_relay();
# end of routing.
}
route[3] {
############################################
# Voicemail specific configuration - begin #
############################################
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if (!t_newtran()) {
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
break;
};
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
if(!vm("/tmp/am_fifo","voicemail")) {
log("couldn't contact announcement server\n");
t_reply("500", "couldn not contact announcement server");
};
break;
};
if(method=="BYE" || method=="CANCEL") {
if(!vm("/tmp/am_fifo","bye")) {
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
break;
};
};
if (method=="CANCEL") {
sl_send_reply("200", "cancels are junked here");
break;
};
sl_send_reply("501", "method not understood here");
}
route[4] {
######################################
# ivr specific configuration - begin #
######################################
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if (!t_newtran()) {
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
break;
};
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
log("**************** vm start - begin ******************\n");
if (uri=~"sip:5000@.*") {
if (!vm("/tmp/am_fifo", "ivr")) {
log("couldn't contact ivr server\n");
t_reply("500", "couldn not contact ivr server");
};
};
log("**************** vm start - end ******************\n");
} else if(method=="BYE"){
log("**************** vm end - begin ******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact ivr\n");
t_reply("500","could not contact ivr");
};
log("**************** vm end - end ******************\n");
};
break;
};
if (method=="CANCEL") {
sl_send_reply("200", "cancels are junked here");
break;
};
sl_send_reply("501", "method not understood here");
}
route[5] {
# ####################################
# conference specific configuration #
# ####################################
if(t_newtran()){
if(method=="INVITE" || method=="BYE" || method=="CANCEL"){
# send a response right at the start to avoid retransmissions
t_reply("100","Trying -- just wait a minute !");
# isdngw only gets activated on invite requests
if(method=="INVITE"){
# filename is defined in sems.conf.
if(uri=~"sip:6000@.*"){
if(!vm("/tmp/am_fifo","conference")){
log("could not contact conference\n");
t_reply("500","could not contact conference");
};
# Allow the announcement module of sems to be used as well.
# This can be useful for testing the conference.
} else if(uri=~"sip:5000@.*"){
if(!vm("/tmp/am_fifo","announcement")){
log("could not contact announcement\n");
t_reply("500","could not contact announcement");
};
# we dont feel responsible for sip addresses of any other kind,
# so send the right error code.
} else {
t_reply("404","Not Found");
};
# stop routing here, the message is now processed by the media server
break;
};
# The following handles the call termination, we must pass these requests
# to the media server as follows. Again make shure the fifo name and permissions
# are set correctly (like im sems.conf).
if((method=="BYE")||(method=="CANCEL")){
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the media server\n");
t_reply("500","could not contact the media server");
};
break;
};
# other methods than INVITE, BYE and CANCEL are not handled by this SIP Server
# so we sent an error message
} else {
log("ERROR: method not supported\n");
t_reply("500", "sorry, method not supported");
};
} else {
# for any reason the transaction could not be created, send error code
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
};
# in isdngw.conf. Don't change this setting.
t_relay();
# end of routing.
}
route[6] {
######################################
# announcement configuration - begin #
######################################
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if (!t_newtran()) {
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
break;
};
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
log("**************** vm start - begin ******************\n");
if (uri=~"sip:7000@.*") {
if (!vm("/tmp/am_fifo", "announcement")) {
log("couldn't contact ivr server\n");
t_reply("500", "couldn not contact announcement");
};
};
log("**************** vm start - end ******************\n");
} else if(method=="BYE"){
log("**************** vm end - begin ******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact annoucement\n");
t_reply("500","could not contact annoucement");
};
log("**************** vm end - end ******************\n");
};
break;
};
if (method=="CANCEL") {
sl_send_reply("200", "cancels are junked here");
break;
};
sl_send_reply("501", "method not understood here");
}
route[7] {
# non-Voip -- just send "off-line"
if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
if (t_newtran()) {
t_reply("404", "Not Found");
acc_db_request("404 missed call", "missed_calls");
};
}
